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	<updated>2026-04-24T06:53:01Z</updated>
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		<title>Router Info</title>
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		<updated>2026-04-08T20:25:12Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Sophos */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;This page contains general information about port forwarding and disabling application layer gateways on particular routers.&lt;br /&gt;
&lt;br /&gt;
[[Router Compatibility|Router Compatibility List]]&lt;br /&gt;
&lt;br /&gt;
= Sonicwall =&lt;br /&gt;
&lt;br /&gt;
{{:Router Info Sonicwall}}&lt;br /&gt;
&lt;br /&gt;
= Mikrotik&amp;lt;br/&amp;gt; =&lt;br /&gt;
&lt;br /&gt;
This router has an ALG that can be disabled with the following command&lt;br /&gt;
&lt;br /&gt;
*&amp;lt;code&amp;gt;/ip firewall service-port disable sip&amp;lt;/code&amp;gt;&lt;br /&gt;
&lt;br /&gt;
The info was found at the following two links [http://wiki.mikrotik.com/wiki/Manual:IP/Services Mikrotik Wiki] [http://forum.mikrotik.com/viewtopic.php?f=13&amp;amp;t=56265 Mikrotik Forum]&lt;br /&gt;
&lt;br /&gt;
= Fortigate&amp;lt;br/&amp;gt; =&lt;br /&gt;
&lt;br /&gt;
==Disable SIP ALG==&lt;br /&gt;
&lt;br /&gt;
====Check the ID of the SIP session helper====&lt;br /&gt;
&lt;br /&gt;
  config system session-helper&lt;br /&gt;
      show&lt;br /&gt;
&lt;br /&gt;
Among the displayed settings will be one similar to the following example:&lt;br /&gt;
&lt;br /&gt;
 edit 13&lt;br /&gt;
       set name sip&lt;br /&gt;
       set protocol 17&lt;br /&gt;
       set port 5060&lt;br /&gt;
     next&lt;br /&gt;
&lt;br /&gt;
Here entry 13 is the one which points to SIP traffic which uses UDP port 5060 for signaling.&lt;br /&gt;
&lt;br /&gt;
In this example, the next commands to remove the corresponding entry would be:&lt;br /&gt;
&lt;br /&gt;
 delete 13&lt;br /&gt;
     end&lt;br /&gt;
&lt;br /&gt;
Note that it is not necessary for the SIP entry to be 13, so cross verify which entry has the sip helper settings.&lt;br /&gt;
&lt;br /&gt;
====Change the default–voip–alg-mode to disable SIP-ALG====&lt;br /&gt;
&lt;br /&gt;
By default, SIP-ALG is enabled, if you run &amp;quot;show full&amp;quot; you will find it is set to &amp;quot;proxy-based&amp;quot; as shown below:&lt;br /&gt;
&lt;br /&gt;
 config system settings&lt;br /&gt;
    set default-voip-alg-mode proxy-based&lt;br /&gt;
    end&lt;br /&gt;
&lt;br /&gt;
Run the following command to disable SIP-ALG (proxy-based) and use SIP-helper (kernel-helper-based):&lt;br /&gt;
&lt;br /&gt;
 config system settings&lt;br /&gt;
    set default-voip-alg-mode kernel-helper-based&lt;br /&gt;
    end&lt;br /&gt;
&lt;br /&gt;
====Either clear sessions, or reboot the FortiGate to ensure changes take effect====&lt;br /&gt;
&lt;br /&gt;
- To clear sessions&lt;br /&gt;
&lt;br /&gt;
Ideally, sessions related to VoIP traffic are deleted.&lt;br /&gt;
&lt;br /&gt;
However, in the case of SIP, this means not only deleting the SIP control sessions but also all sessions opened to handle the audio (RTP) traffic.&lt;br /&gt;
Knowing the port-range used for the audio traffic, sessions clear can be selected by first applying a filter as follows:&lt;br /&gt;
&lt;br /&gt;
 diagnose system session filter ...&lt;br /&gt;
&lt;br /&gt;
The command to clear sessions applies to ALL sessions unless a filter is applied, and therefore will interrupt all traffic!&lt;br /&gt;
&lt;br /&gt;
 diagnose system session clear&lt;br /&gt;
&lt;br /&gt;
-  Alternatively, reboot the FortiGate using either GUI or CLI. The CLI command is:&lt;br /&gt;
&lt;br /&gt;
 execute reboot&lt;br /&gt;
&lt;br /&gt;
==Disable NAT on IPv4 Policy==&lt;br /&gt;
&lt;br /&gt;
A common mistake is to create an IPv4 policy for UDP ports 5060 and UDP ports 10000-20000 and to leave the NAT switch on.&amp;lt;br/&amp;gt;&lt;br /&gt;
This makes all incoming traffic on those ports appear to come directly from the private IP address of the Fortigate.&amp;lt;br/&amp;gt;&lt;br /&gt;
If you see this behavior, disable the NAT switch on the IPv4 policy.&amp;lt;br/&amp;gt;&lt;br /&gt;
&lt;br /&gt;
[[File:fortigatenat.png]]&lt;br /&gt;
&lt;br /&gt;
==Ensure Outbound Traffic is Allowed from PBX==&lt;br /&gt;
&lt;br /&gt;
Many Fortigate configurations I've seen have most outbound traffic blocked.&amp;lt;br/&amp;gt;&lt;br /&gt;
In these configurations you will need to create an outbound rule allowing the PBX to talk to our SIP servers from source port 5060 UDP and 10000-20000 UDP.&amp;lt;br/&amp;gt;&lt;br /&gt;
Ideally you would allow the PBX to go outbound on any port it chooses so it can check for updates, provide web and SSH access, etc.&amp;lt;br/&amp;gt;&lt;br /&gt;
&lt;br /&gt;
== Fortigate #2 ==&lt;br /&gt;
'''For Fortigate Firewalls running OSv6 using IPitomy Cloud PBX'''&lt;br /&gt;
&lt;br /&gt;
After testing the Fortigate series firewalls and working with Fortigate support, we have come up with these modifications to Fortigate Firewalls to ensure they work well with IPitomy Cloud PBX’s.&lt;br /&gt;
&lt;br /&gt;
'''If the firewall is not correctly configured, users with these devices will run into the following issues using a Fortigate:'''&lt;br /&gt;
&lt;br /&gt;
1.   Dropped calls&lt;br /&gt;
&lt;br /&gt;
2.   One way or no way audio &lt;br /&gt;
&lt;br /&gt;
3.   Potential device registration issues&lt;br /&gt;
&lt;br /&gt;
4.   Duplicate SIP Ports and port shuffling&lt;br /&gt;
&lt;br /&gt;
To mitigate some of these issues, '''Strict Register''' should be ''disabled (outlined below)'' to stop all phones from using a pinhole through port 65476 (external) and 5060 (internal), aka ALG management. After this is complete if issues persist, set the local SIP ports on each phone to unique port assignments.&lt;br /&gt;
&lt;br /&gt;
'''Delete SIP Firewall'''&lt;br /&gt;
&lt;br /&gt;
1.   '''In the Command Line Interface (CLI) run the following commands:'''&lt;br /&gt;
&lt;br /&gt;
·        ''config system session-helper''&lt;br /&gt;
&lt;br /&gt;
·        ''show''&lt;br /&gt;
[[File:Fort1.png|frameless]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
2.   Notice that edit 13 contains SIP.&lt;br /&gt;
&lt;br /&gt;
3.   '''Enter the following commands:'''&lt;br /&gt;
&lt;br /&gt;
·        ''delete 13''&lt;br /&gt;
&lt;br /&gt;
·        ''end''&lt;br /&gt;
&lt;br /&gt;
'''Disable SIP Helper'''&lt;br /&gt;
&lt;br /&gt;
1.   Default setting is proxy, we want to turn that off&lt;br /&gt;
&lt;br /&gt;
2.   '''In the Command Line Interface (CLI) run the following commands:'''&lt;br /&gt;
&lt;br /&gt;
·        ''config system settings''&lt;br /&gt;
&lt;br /&gt;
·        ''set default-voip-alg-mode kernel-helper-based''&lt;br /&gt;
&lt;br /&gt;
·        ''set sip-helper disable (or) set sip-expectation disable (depending on FortiOS)''&lt;br /&gt;
&lt;br /&gt;
·        ''set sip-nat-trace disable''&lt;br /&gt;
&lt;br /&gt;
·        ''end''&lt;br /&gt;
&lt;br /&gt;
''If using the CLI Console:''&lt;br /&gt;
&lt;br /&gt;
[[File:Fort2.png|frameless]]&lt;br /&gt;
&lt;br /&gt;
3.   '''Reboot the router using the web GUI under Status, or in the CLI with the following command:'''&lt;br /&gt;
&lt;br /&gt;
·        ''execute reboot'' &lt;br /&gt;
&lt;br /&gt;
'''Configure Traffic Shaping and VoIP'''&lt;br /&gt;
&lt;br /&gt;
1.   In the web GUI, go to '''System''' &amp;gt; '''Feature Select''' &amp;gt; '''Additional Features'''.&lt;br /&gt;
&lt;br /&gt;
[[File:Fort3.png|frameless]]&lt;br /&gt;
&lt;br /&gt;
2.   Toggle '''Traffic Shaping''' and '''VoIP''' on.&lt;br /&gt;
&lt;br /&gt;
3.   Click '''Apply'''.&lt;br /&gt;
&lt;br /&gt;
'''Disable Strict Register'''&lt;br /&gt;
&lt;br /&gt;
Strict Register forces VoIP devices through a pinhole at port 65476 and will cause duplicate porting to occur.&lt;br /&gt;
&lt;br /&gt;
'''To disable this setting run the following command in the Command Line Interface (CLI):'''&lt;br /&gt;
&lt;br /&gt;
·        ''config voip profile''&lt;br /&gt;
&lt;br /&gt;
·        ''edit &amp;lt;Profile_name&amp;gt;''&lt;br /&gt;
&lt;br /&gt;
·        ''config sip''&lt;br /&gt;
&lt;br /&gt;
·        ''set strict-register disable''&lt;br /&gt;
&lt;br /&gt;
·        ''end''&lt;br /&gt;
&lt;br /&gt;
''Note: The VoIP profile name can be found under Security Profile &amp;gt; VoIP &amp;gt; default. Please note if these settings do not persist through a reboot a factory reset or other troubleshooting steps may be needed on the Fortigate itself with Fortigate support.''&lt;br /&gt;
[[File:Fort4.png|none|thumb]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Create SIP PBX Objects'''&lt;br /&gt;
&lt;br /&gt;
1.  In the web GUI, go to '''Policy &amp;amp; Objects'''.&lt;br /&gt;
&lt;br /&gt;
2.  Find '''Addresses'''.&lt;br /&gt;
&lt;br /&gt;
3.  Click '''Create New''', then click '''Address'''.&lt;br /&gt;
[[File:Fort5.png|none|thumb]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
4.  You will need to add each subnet in the format xxx.xx.xx.x/xx.&lt;br /&gt;
&lt;br /&gt;
5.  Do this for each of the related SIP PBX subnets that should be allowed. &lt;br /&gt;
&lt;br /&gt;
6.  Find the IP addresses the PBX’s can use and enter them like this: 216.15.137.21/21, or by range:  52.5.220.123-52.5.220.125 and 165.227.0.0/16&lt;br /&gt;
&lt;br /&gt;
1.   Allow the IP address for the specific IPitomy PBX, you can ping the FQDN or check in System &amp;gt; Networking for the public IP address.&lt;br /&gt;
&lt;br /&gt;
1.   You can enter it as a FQDN also.&lt;br /&gt;
&lt;br /&gt;
2.   You may want to verify these IP addresses with IPitomy support.&lt;br /&gt;
&lt;br /&gt;
3.   Allow SIP, IP’s: &lt;br /&gt;
&lt;br /&gt;
1.   52.5.220.123-52.5.220.124&lt;br /&gt;
&lt;br /&gt;
2.   54.200.236.200/30&lt;br /&gt;
&lt;br /&gt;
4.   Allow proxy for soft phones: &lt;br /&gt;
&lt;br /&gt;
1.   165.227.0.0/16&lt;br /&gt;
&lt;br /&gt;
2.   167.99.0.0/16&lt;br /&gt;
&lt;br /&gt;
3.   159.65.0.0/16&lt;br /&gt;
&lt;br /&gt;
'''Group the Telecom Networks'''&lt;br /&gt;
&lt;br /&gt;
1.   In the web GUI, go to '''Policy &amp;amp; Objects'''.&lt;br /&gt;
&lt;br /&gt;
2.   Select '''Objects''', then '''Addresses'''.&lt;br /&gt;
&lt;br /&gt;
3.   Click '''Create New''', then click '''Address Group'''.&lt;br /&gt;
&lt;br /&gt;
4.   Create a '''Group Name'''.&lt;br /&gt;
&lt;br /&gt;
5.   Click '''Members''', click each subnet, then click '''OK'''.&lt;br /&gt;
&lt;br /&gt;
[[File:Fort6.png|frameless]]&lt;br /&gt;
&lt;br /&gt;
'''Set High-Priority Traffic Guarantee'''&lt;br /&gt;
&lt;br /&gt;
1.   In the web GUI, go to '''Policy &amp;amp; Objects'''.&lt;br /&gt;
&lt;br /&gt;
2.   Select '''Traffic Shapers'''.&lt;br /&gt;
&lt;br /&gt;
3.   Edit the existing '''High Priority Traffic Shaper'''.&lt;br /&gt;
&lt;br /&gt;
4.   Set '''Type''' to '''Shared'''.&lt;br /&gt;
&lt;br /&gt;
5.   Set '''Apply Shaper''' to '''Per Policy'''.&lt;br /&gt;
&lt;br /&gt;
6.   Set '''Traffic Priority''' to '''High'''.&lt;br /&gt;
&lt;br /&gt;
7.   Check '''Max Bandwidth''' and set to '''1048576 Kb/s'''.&lt;br /&gt;
&lt;br /&gt;
8.   Check '''Guaranteed Bandwidth''' and set to '''1000 Kb/s'''.&lt;br /&gt;
&lt;br /&gt;
9.   Click '''OK'''.&lt;br /&gt;
&lt;br /&gt;
[[File:Fort7.png|frameless]]&lt;br /&gt;
&lt;br /&gt;
'''Create a New Policy'''&lt;br /&gt;
&lt;br /&gt;
1.   In the web GUI, go to '''Policy &amp;amp; Objects''' &amp;gt; '''Policy'''.&lt;br /&gt;
&lt;br /&gt;
2.   Select '''IPv4'''.&lt;br /&gt;
&lt;br /&gt;
3.   Create a new policy.&lt;br /&gt;
&lt;br /&gt;
4.   '''Set the following options:'''&lt;br /&gt;
&lt;br /&gt;
·        Incoming Interface: Internal&lt;br /&gt;
&lt;br /&gt;
·        Source Address: All&lt;br /&gt;
&lt;br /&gt;
·        Outgoing Interface: WAN&lt;br /&gt;
&lt;br /&gt;
·        Destination Address: Whatever you named the address Group&lt;br /&gt;
&lt;br /&gt;
·        Service: All&lt;br /&gt;
&lt;br /&gt;
·        Service: SIP, RTSP&lt;br /&gt;
&lt;br /&gt;
5.   Click '''OK'''.&lt;br /&gt;
&lt;br /&gt;
[[File:Fort8.png|frameless]] &lt;br /&gt;
&lt;br /&gt;
6.   Add additional Ports.&lt;br /&gt;
&lt;br /&gt;
#* You may need to add IAX2, the      Asterisk Protocol and Queue Manager to services and include them in the      above services&lt;br /&gt;
#* IAX2 is port 4569 TCP &amp;amp;      UDP&lt;br /&gt;
#* Queue Manager is port 5048 TCP      &amp;amp; UDP&lt;br /&gt;
&lt;br /&gt;
[[File:Fort9.png|frameless]] &lt;br /&gt;
&lt;br /&gt;
'''Arrange Policy'''&lt;br /&gt;
&lt;br /&gt;
1.   In the web GUI, go to '''Policy &amp;amp; Objects''' &amp;gt; IPv4 '''Policy'''.&lt;br /&gt;
&lt;br /&gt;
2.   Double-click the new policy you just made.&lt;br /&gt;
&lt;br /&gt;
3.   Drag and drop the '''new''' policy to the top spot.&lt;br /&gt;
&lt;br /&gt;
[[File:Fort10.png|frameless]]&lt;br /&gt;
&lt;br /&gt;
'''Verify Traffic'''&lt;br /&gt;
&lt;br /&gt;
Start a call and download some large traffic &amp;lt;nowiki&amp;gt;https://www.nasa.gov/content/ultra-high-definition-video-gallery&amp;lt;/nowiki&amp;gt; (e.g.,  shown below) or use a tool like iperf (&amp;lt;nowiki&amp;gt;https://iperf.fr/&amp;lt;/nowiki&amp;gt; )&lt;br /&gt;
&lt;br /&gt;
[[File:Fort11.png|frameless]]&lt;br /&gt;
&lt;br /&gt;
'''Review Traffic Shaper Monitor'''&lt;br /&gt;
&lt;br /&gt;
1.   In the web GUI, go to '''Policy &amp;amp; Objects''' &amp;gt; '''Monitor'''.&lt;br /&gt;
&lt;br /&gt;
2.   Select '''Traffic Shaper Monitor'''.&lt;br /&gt;
&lt;br /&gt;
3.   Note that since the default traffic is left alone, it doesn’t show in the Traffic Shaper Monitor. Only the reserved traffic displays, in this case high-priority.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Additional Information:''' &lt;br /&gt;
&lt;br /&gt;
Firmware updates may re-enable some system settings. After a firmware update if you begin to experience issues with the phones again, verify the firewall still has the proper configuration. &lt;br /&gt;
&lt;br /&gt;
'''To validate the settings implemented are written successfully to the configuration file, download a back up of the existing configuration and verify the following information has been written:'''&lt;br /&gt;
&lt;br /&gt;
config voip profile&lt;br /&gt;
&lt;br /&gt;
    edit &amp;quot;default&amp;quot;&lt;br /&gt;
&lt;br /&gt;
        set comment &amp;quot;Default VoIP profile.&amp;quot;&lt;br /&gt;
&lt;br /&gt;
        config sip&lt;br /&gt;
&lt;br /&gt;
            set strict-register disable&lt;br /&gt;
&lt;br /&gt;
            set register-rate 300&lt;br /&gt;
&lt;br /&gt;
           set invite-rate 300&lt;br /&gt;
&lt;br /&gt;
    end&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
config firewall policy&lt;br /&gt;
&lt;br /&gt;
           edit&lt;br /&gt;
&lt;br /&gt;
show&lt;br /&gt;
&lt;br /&gt;
''make note of the numbers of the policy(s) used by Voip, usually the main policy and the SIP policy''&lt;br /&gt;
&lt;br /&gt;
           edit (policy number)&lt;br /&gt;
&lt;br /&gt;
                       set inspection-mode flow&lt;br /&gt;
&lt;br /&gt;
           end&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
config system settings&lt;br /&gt;
&lt;br /&gt;
    set sip-nat-trace disable&lt;br /&gt;
&lt;br /&gt;
    set default-voip-alg-mode kernel-helper-based&lt;br /&gt;
&lt;br /&gt;
    set gui-voip-profile enable&lt;br /&gt;
&lt;br /&gt;
   end&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To extend the time for ports and prevent one-way audio after 30 to 45 seconds:&lt;br /&gt;
&lt;br /&gt;
config sys global&lt;br /&gt;
&lt;br /&gt;
set udp-idle-timer 300&lt;br /&gt;
&lt;br /&gt;
end&lt;br /&gt;
&lt;br /&gt;
== Fortigate #3 ==&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''For Fortigate Firewalls running OSv7 using IPitomy Cloud PBX'''&lt;br /&gt;
&lt;br /&gt;
After testing the Fortigate series firewalls and working with Fortigate support, we have come up with these modifications to Fortigate Firewalls to ensure they work well with IPitomy Cloud PBX’s.&lt;br /&gt;
&lt;br /&gt;
'''If the firewall is not correctly configured, users with these devices will run into the following issues using a Fortigate:'''&lt;br /&gt;
&lt;br /&gt;
1.   Dropped calls&lt;br /&gt;
&lt;br /&gt;
2.   One way or no way audio &lt;br /&gt;
&lt;br /&gt;
3.   Poor Voice Quality/Clicking&lt;br /&gt;
&lt;br /&gt;
4.   Potential device registration issues&lt;br /&gt;
&lt;br /&gt;
5.   Duplicate SIP Ports and port shuffling which result in poor quality voice.&lt;br /&gt;
&lt;br /&gt;
To mitigate some of these issues, '''Strict Register''' should be ''disabled (outlined below)'' to stop all phones from using a pinhole through port 65476 (external) and 5060 (internal), aka ALG management.  That may be all you need to do with some PBX’s, especially premise.  These instructions are designed for CLOUD IPitomy PBX’s. &lt;br /&gt;
&lt;br /&gt;
'''''NOTE:'''''  With FortiOS version 7 comes a change to profiles.  It used to be that there was a default Security Policy for each Security Policy, and you could change it.  With version 7 you cannot change the default policy, you must make a clone or use the premade clone called “Base”.&lt;br /&gt;
&lt;br /&gt;
'''''ADDITIONALLY!'''''  When editing Firewall policies, to add security profiles, the policy Inspection Mode MUST be set to Proxy-Based, otherwise you don’t see any Security policy except Default.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
= Cisco Pix 506/501/515 and Cisco ASA&amp;lt;br/&amp;gt; =&lt;br /&gt;
&amp;lt;div dir=&amp;quot;ltr&amp;quot; data-font-name=&amp;quot;g_font_p0_1&amp;quot; data-canvas-width=&amp;quot;271.1786496402639&amp;quot;&amp;gt;This is for Pix 506/501/515 but it should work with any Cisco Pix, and possibly other Cisco&amp;lt;br/&amp;gt;&amp;lt;/div&amp;gt;&amp;lt;div dir=&amp;quot;ltr&amp;quot; data-font-name=&amp;quot;g_font_p0_1&amp;quot; data-canvas-width=&amp;quot;48.802877652486835&amp;quot;&amp;gt;routers.&amp;lt;/div&amp;gt;&lt;br /&gt;
#access-list 101 permit udp any host 64.238.XXX.XXX range 10000 20000&amp;lt;br/&amp;gt;(Note: Replace 64.238.XXX.XXX with your public IP assigned to be forwarded to the IPitomy PBX)&lt;br /&gt;
#access-list 101 permit tcp any host 64.238.XXX.XXX range 10000 20000&amp;lt;br/&amp;gt;(Note: Replace 64.238.XXX.XXX with your public IP assigned to be forwarded to the IPitomy PBX)&lt;br /&gt;
#static (inside,outside) 64.238.XXX.XX 172.16.2.129 netmask 255.255.255.255 0 0&amp;lt;br/&amp;gt;(Note: Replace 64.238.XXX.XXX with users public IP, replace the 172.16.2.129 with users private IP that is assigned to the IPitomy PBX)&lt;br /&gt;
#no fixup protocol sip 5060&lt;br /&gt;
#no fixup protocol sip udp 5060&lt;br /&gt;
&lt;br /&gt;
= Adtran =&lt;br /&gt;
&lt;br /&gt;
From a recent interaction with an AdTran tech, it was shown to us there is a setting for &amp;quot;proxy transparency&amp;quot; that needs to be enabled in order for all of the SIP traffic to pass unhindered. This was when the Adtran was the routing device at the remote site, but likely would need to be enabled when the Adtran is at the PBX site. Its worth trying for sure.&lt;br /&gt;
&lt;br /&gt;
= PepLink =&lt;br /&gt;
&lt;br /&gt;
Here is a document sent to a dealer from PepLink regarding configuration settings that may be required for Remote SIP to function properly:&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:Peplink Config.pdf|File:Peplink Config.pdf]]&lt;br /&gt;
&lt;br /&gt;
= FIOS ActionTec =&lt;br /&gt;
&lt;br /&gt;
We have found the following article that outlines some possible configurations that are available on the Actiontec Modem/Router combo that FIOS is installing.  This gives some options on ways to configure to optimize VoIP and SIP traffic passing to remote.&lt;br /&gt;
&lt;br /&gt;
http://www.dslreports.com/faq/verizonfios/3.0_Networking#16077&lt;br /&gt;
&lt;br /&gt;
= Comcast Modem =&lt;br /&gt;
&lt;br /&gt;
We have received some information from our dealers that if your site has a Comcast modem/router, you should request a SMC and not a Linksys, as the reports are that the SMC handles VoIP more consistently.  Additionally, there may be issues with Comcast modem/routers ability to handle multiple concurrent NAT sessions, limiting the number of remote phones you can install at a remote site.&lt;br /&gt;
&lt;br /&gt;
= Sophos =&lt;br /&gt;
&lt;br /&gt;
Some Sophos models have a hidden SIP module that is not in any way indicated, nor accessible, from the webgui.  It must be disabled from the command line console.  If left enabled, it attempts to override any rules you may have in place for sip/rtp traffic and can result in one-way audio, and other issues with calls successfully connecting.&lt;br /&gt;
&lt;br /&gt;
== TP-link ER605 ==&lt;br /&gt;
[[File:TPlink.jpg|border|left|thumb]]&lt;br /&gt;
Apparently you need to have h323 alg turned on for voip to work on the old Ipitomy phones.&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=File:TPlink.jpg&amp;diff=5185</id>
		<title>File:TPlink.jpg</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=File:TPlink.jpg&amp;diff=5185"/>
		<updated>2026-04-08T20:24:32Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;jw&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=USB_Install_Dongle&amp;diff=5184</id>
		<title>USB Install Dongle</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=USB_Install_Dongle&amp;diff=5184"/>
		<updated>2026-03-17T17:58:42Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Using the USB Install Dongle */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;If you are advised by Tech Support to perform a field install of PBX software, these are the instructions to create the USB dongle and how to use that to install the software on the PBX. Once completed, you will most likely need to upgrade the PBX to current software version via the normal process.&lt;br /&gt;
&lt;br /&gt;
== Creating a USB Install Dongle ==&lt;br /&gt;
&lt;br /&gt;
You'll find both the LiveUSBCreator.zip file here: [https://www.dropbox.com/sh/hsdsk0lr00rnlr5/AABTL5y38oBfu3vKvBhrodGba/USB_Disk/install_disk?dl=0&amp;amp;preview=LiveUSBCreator.zip Live USB Creator] The version 6 install ISO can be found here: [https://www.dropbox.com/sh/hsdsk0lr00rnlr5/AADLN4zNL0MSNy7SAjfXhq12a/IPitomy/IP1100%2B?dl=0&amp;amp;preview=IPitomyPBXReinstall.iso Version 6 Install Image]&amp;amp;nbsp;(NOTE: Installing version 6 REQUIRES a 64-bit compatible system, please check with tech support if you are unsure about your pbx).&lt;br /&gt;
&lt;br /&gt;
When you click on the links, they will take you to a Dropbox.  There you will navigate to the right side of the screen and click on the download button to download the files.&lt;br /&gt;
&lt;br /&gt;
[[File:USB999.png|frameless|600x600px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:USB25555.png|frameless|600x600px]]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br /&amp;gt;Download the file, LiveUSBCreator.zip and extract it's contents. This will create a folder called LiveUSBCreator Download the file, IPitomyPBXReinstall.iso. Put this file in the newly created LiveUSBCreator folder Put a formated (FAT32), empty 1GB USB drive in the system.  Enter the folder LiveUSBCreator.  Right click on file: liveusb-creator application, and Run as Administrator.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Usb2.png|frame|alt=|none]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:USB.png|frame|alt=|none]]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:001.jpg|File:001.jpg]]&lt;br /&gt;
&lt;br /&gt;
The USB drive letter should be visible in the &amp;quot;Target Device&amp;quot; box. If not, then select it from the list. Click the [Browse] button under &amp;quot;Use existing Live CD&amp;quot; Browse to and select &amp;quot; Install .iso&amp;quot; Click [Open]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:002.jpg|File:002.jpg]]&lt;br /&gt;
&lt;br /&gt;
Click [Create Live USB]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:003.jpg|File:003.jpg]]&lt;br /&gt;
&lt;br /&gt;
After a few minutes, the Install USB will have been created:&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:004.jpg|File:004.jpg]]&lt;br /&gt;
&lt;br /&gt;
Click the [X] in the upper right corner to close the program and properly eject the USB drive from the system&lt;br /&gt;
&lt;br /&gt;
== Using the USB Install Dongle ==&lt;br /&gt;
&lt;br /&gt;
== Once you are finished you may shutdown the pbx you are going to install the operating system on.  Insert the USB drive into the pbx and power it back on with a USB keyboard and a monitor atttached to it.  May sure you are connected to the internet.  Once the pbx boots up you will see a menu for downloading the OS. ==&lt;br /&gt;
&amp;lt;big&amp;gt;When reinstalling a PBX or upgrading to a new OS from a USB dongle, there are two things that need to be done after the system is relicensed.&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;First,&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;Go to System -&amp;gt; VLAN&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;Hit 'Save' and 'Apply Changes'&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;&amp;lt;br /&amp;gt;&lt;br /&gt;
Then do the same in Phone Global&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;Go to PBX Settings -&amp;gt; Phone Global&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;Hit 'Save' and 'Apply Changes'&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;&amp;lt;br /&amp;gt;&lt;br /&gt;
It does not matter whether or not you are using VLAN or made any changes on Phone Global.&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;&amp;lt;br /&amp;gt;&lt;br /&gt;
The VLAN Save configures the DNS Client so that names resolve.&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;The Phone Global Save initializes various passwords so they can be written to the phone configs.&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Tech Bulletin 2011-009 - IPitomy ERD v 2.0.pdf|File:Tech Bulletin 2011-009 - IPitomy ERD v 2.0.pdf]]&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=USB_Install_Dongle&amp;diff=5183</id>
		<title>USB Install Dongle</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=USB_Install_Dongle&amp;diff=5183"/>
		<updated>2026-03-17T17:57:20Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Using the USB Install Dongle */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;If you are advised by Tech Support to perform a field install of PBX software, these are the instructions to create the USB dongle and how to use that to install the software on the PBX. Once completed, you will most likely need to upgrade the PBX to current software version via the normal process.&lt;br /&gt;
&lt;br /&gt;
== Creating a USB Install Dongle ==&lt;br /&gt;
&lt;br /&gt;
You'll find both the LiveUSBCreator.zip file here: [https://www.dropbox.com/sh/hsdsk0lr00rnlr5/AABTL5y38oBfu3vKvBhrodGba/USB_Disk/install_disk?dl=0&amp;amp;preview=LiveUSBCreator.zip Live USB Creator] The version 6 install ISO can be found here: [https://www.dropbox.com/sh/hsdsk0lr00rnlr5/AADLN4zNL0MSNy7SAjfXhq12a/IPitomy/IP1100%2B?dl=0&amp;amp;preview=IPitomyPBXReinstall.iso Version 6 Install Image]&amp;amp;nbsp;(NOTE: Installing version 6 REQUIRES a 64-bit compatible system, please check with tech support if you are unsure about your pbx).&lt;br /&gt;
&lt;br /&gt;
When you click on the links, they will take you to a Dropbox.  There you will navigate to the right side of the screen and click on the download button to download the files.&lt;br /&gt;
&lt;br /&gt;
[[File:USB999.png|frameless|600x600px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:USB25555.png|frameless|600x600px]]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br /&amp;gt;Download the file, LiveUSBCreator.zip and extract it's contents. This will create a folder called LiveUSBCreator Download the file, IPitomyPBXReinstall.iso. Put this file in the newly created LiveUSBCreator folder Put a formated (FAT32), empty 1GB USB drive in the system.  Enter the folder LiveUSBCreator.  Right click on file: liveusb-creator application, and Run as Administrator.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Usb2.png|frame|alt=|none]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:USB.png|frame|alt=|none]]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:001.jpg|File:001.jpg]]&lt;br /&gt;
&lt;br /&gt;
The USB drive letter should be visible in the &amp;quot;Target Device&amp;quot; box. If not, then select it from the list. Click the [Browse] button under &amp;quot;Use existing Live CD&amp;quot; Browse to and select &amp;quot; Install .iso&amp;quot; Click [Open]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:002.jpg|File:002.jpg]]&lt;br /&gt;
&lt;br /&gt;
Click [Create Live USB]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:003.jpg|File:003.jpg]]&lt;br /&gt;
&lt;br /&gt;
After a few minutes, the Install USB will have been created:&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:004.jpg|File:004.jpg]]&lt;br /&gt;
&lt;br /&gt;
Click the [X] in the upper right corner to close the program and properly eject the USB drive from the system&lt;br /&gt;
&lt;br /&gt;
== Using the USB Install Dongle ==&lt;br /&gt;
&lt;br /&gt;
== Once you are finished you may shutdown the pbx you are going to install the operating system on.  Insert the USB drive into the pbx and power it back on with a USB keyboard and a monitor atttached to it.  May sure you are connected to the internet.  Once the pbx boots up you will see a menu for downloading the OS. ==&lt;br /&gt;
&amp;lt;big&amp;gt;When reinstalling a PBX or upgrading to a new OS from a USB dongle, there are two things that need to be done after the system is relicensed.&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;First,&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;Go to System -&amp;gt; VLAN&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;Hit 'Save' and 'Apply Changes'&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;&amp;lt;br /&amp;gt;&lt;br /&gt;
Then do the same in Phone Global&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;Go to PBX Settings -&amp;gt; Phone Global&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;Hit 'Save' and 'Apply Changes'&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;&amp;lt;br /&amp;gt;&lt;br /&gt;
It does not matter whether or not you are using VLAN or made any changes on Phone Global.&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;&amp;lt;br /&amp;gt;&lt;br /&gt;
The VLAN Save configures the DNS Client so that names resolve.&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;The Phone Global Save initializes various passwords so they can be written to the phone configs.&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
[[File:Tech Bulletin 2011-009 - IPitomy ERD v 2.0.pdf|File:Tech Bulletin 2011-009 - IPitomy ERD v 2.0.pdf]]&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=USB_Install_Dongle&amp;diff=5182</id>
		<title>USB Install Dongle</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=USB_Install_Dongle&amp;diff=5182"/>
		<updated>2026-03-17T17:56:56Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Using the USB Install Dongle */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;If you are advised by Tech Support to perform a field install of PBX software, these are the instructions to create the USB dongle and how to use that to install the software on the PBX. Once completed, you will most likely need to upgrade the PBX to current software version via the normal process.&lt;br /&gt;
&lt;br /&gt;
== Creating a USB Install Dongle ==&lt;br /&gt;
&lt;br /&gt;
You'll find both the LiveUSBCreator.zip file here: [https://www.dropbox.com/sh/hsdsk0lr00rnlr5/AABTL5y38oBfu3vKvBhrodGba/USB_Disk/install_disk?dl=0&amp;amp;preview=LiveUSBCreator.zip Live USB Creator] The version 6 install ISO can be found here: [https://www.dropbox.com/sh/hsdsk0lr00rnlr5/AADLN4zNL0MSNy7SAjfXhq12a/IPitomy/IP1100%2B?dl=0&amp;amp;preview=IPitomyPBXReinstall.iso Version 6 Install Image]&amp;amp;nbsp;(NOTE: Installing version 6 REQUIRES a 64-bit compatible system, please check with tech support if you are unsure about your pbx).&lt;br /&gt;
&lt;br /&gt;
When you click on the links, they will take you to a Dropbox.  There you will navigate to the right side of the screen and click on the download button to download the files.&lt;br /&gt;
&lt;br /&gt;
[[File:USB999.png|frameless|600x600px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:USB25555.png|frameless|600x600px]]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br /&amp;gt;Download the file, LiveUSBCreator.zip and extract it's contents. This will create a folder called LiveUSBCreator Download the file, IPitomyPBXReinstall.iso. Put this file in the newly created LiveUSBCreator folder Put a formated (FAT32), empty 1GB USB drive in the system.  Enter the folder LiveUSBCreator.  Right click on file: liveusb-creator application, and Run as Administrator.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Usb2.png|frame|alt=|none]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:USB.png|frame|alt=|none]]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:001.jpg|File:001.jpg]]&lt;br /&gt;
&lt;br /&gt;
The USB drive letter should be visible in the &amp;quot;Target Device&amp;quot; box. If not, then select it from the list. Click the [Browse] button under &amp;quot;Use existing Live CD&amp;quot; Browse to and select &amp;quot; Install .iso&amp;quot; Click [Open]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:002.jpg|File:002.jpg]]&lt;br /&gt;
&lt;br /&gt;
Click [Create Live USB]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:003.jpg|File:003.jpg]]&lt;br /&gt;
&lt;br /&gt;
After a few minutes, the Install USB will have been created:&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:004.jpg|File:004.jpg]]&lt;br /&gt;
&lt;br /&gt;
Click the [X] in the upper right corner to close the program and properly eject the USB drive from the system&lt;br /&gt;
&lt;br /&gt;
== Using the USB Install Dongle ==&lt;br /&gt;
&lt;br /&gt;
== Once you are finished you may shutdown the pbx you are going to install the operating system on.  Insert the USB drive into the pbx and power it back on with a USB keyboard and a monitor atttached to it.  May sure you are connected to the internet.  Once the pbx boots up you will see a menu for downloading the OS. ==&lt;br /&gt;
When reinstalling a PBX or upgrading to a new OS from a USB dongle, there are two things that need to be done after the system is relicensed.&lt;br /&gt;
&lt;br /&gt;
First,&lt;br /&gt;
&lt;br /&gt;
Go to System -&amp;gt; VLAN&lt;br /&gt;
&lt;br /&gt;
Hit 'Save' and 'Apply Changes'&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Then do the same in Phone Global&lt;br /&gt;
&lt;br /&gt;
Go to PBX Settings -&amp;gt; Phone Global&lt;br /&gt;
&lt;br /&gt;
Hit 'Save' and 'Apply Changes'&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
It does not matter whether or not you are using VLAN or made any changes on Phone Global.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
The VLAN Save configures the DNS Client so that names resolve.&lt;br /&gt;
&lt;br /&gt;
The Phone Global Save initializes various passwords so they can be written to the phone configs.&lt;br /&gt;
&lt;br /&gt;
[[File:Tech Bulletin 2011-009 - IPitomy ERD v 2.0.pdf|File:Tech Bulletin 2011-009 - IPitomy ERD v 2.0.pdf]]&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=Port_Forwarding&amp;diff=5175</id>
		<title>Port Forwarding</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=Port_Forwarding&amp;diff=5175"/>
		<updated>2025-08-08T12:08:36Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Port Forwarding */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;__NOTITLE__&lt;br /&gt;
[[Category: Router Configuration]]&lt;br /&gt;
&lt;br /&gt;
== Port Forwarding&amp;lt;br/&amp;gt; ==&lt;br /&gt;
&lt;br /&gt;
The following table outlines the port forwarding information in the router that maps public IP addresses to internal IP addresses. Port forwarding must be configured to utilize features such as remote phones, SIP Providers, remote administration and branch office. IPitomy port forwarding requirements are specified below. Note you should only forward ports you intend to use.&lt;br /&gt;
&lt;br /&gt;
;Remote Phones&amp;amp;nbsp;&lt;br /&gt;
:SIP + RTP&lt;br /&gt;
;Web Access to PBX (do not forward without changing password)&amp;amp;nbsp;&lt;br /&gt;
:HTTP/HTTPS&lt;br /&gt;
;Branch Office Networking&amp;amp;nbsp;&lt;br /&gt;
:IAX2&lt;br /&gt;
;Text Messaging (via eJabbard - NOT SMS)&lt;br /&gt;
:XMPP&lt;br /&gt;
;Tech Support Access&amp;amp;nbsp;&lt;br /&gt;
:SSH - you should always forward this before you contact Technical Support&lt;br /&gt;
&lt;br /&gt;
{| class=&amp;quot;wikitable&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
! Application&lt;br /&gt;
! Port&lt;br /&gt;
! Protocol&lt;br /&gt;
! Transport Protocol&lt;br /&gt;
! LAN IP Address&lt;br /&gt;
!Assigned WAN IP&lt;br /&gt;
Addresses&lt;br /&gt;
|-&lt;br /&gt;
| Remote Management&amp;lt;br/&amp;gt;&lt;br /&gt;
| 80&lt;br /&gt;
| HTTP&lt;br /&gt;
| TCP&lt;br /&gt;
| IP Address of PBX&lt;br /&gt;
|Not Assigned&lt;br /&gt;
|-&lt;br /&gt;
| Secure Remote Management&amp;lt;br/&amp;gt;&lt;br /&gt;
| 443&amp;lt;br/&amp;gt;&lt;br /&gt;
| HTTPS&lt;br /&gt;
| TCP&lt;br /&gt;
| IP Address of PBX&amp;lt;br/&amp;gt;&lt;br /&gt;
|Not Assigned&lt;br /&gt;
|-&lt;br /&gt;
| SSH (IPitomy Support)&amp;lt;br/&amp;gt;&lt;br /&gt;
| 22&lt;br /&gt;
| SSH&amp;lt;br/&amp;gt;&lt;br /&gt;
| TCP&lt;br /&gt;
| IP Address of PBX&lt;br /&gt;
|47.206.59.221, &lt;br /&gt;
74.93.34.251&lt;br /&gt;
|-&lt;br /&gt;
| SIP Control&lt;br /&gt;
| 5060&lt;br /&gt;
| SIP&lt;br /&gt;
| UDP&lt;br /&gt;
| IP Address of PBX&lt;br /&gt;
|52.5.220.123     (SIP EAST)&lt;br /&gt;
54.200.236.200 (SIP WEST)&lt;br /&gt;
|-&lt;br /&gt;
| SIP RTP (Voice)&lt;br /&gt;
| 10000-20000&lt;br /&gt;
| RTP&lt;br /&gt;
| UDP + TCP&lt;br /&gt;
| IP Address of PBX&lt;br /&gt;
|52.5.220.123     (SIP EAST)&lt;br /&gt;
&lt;br /&gt;
54.200.236.200 (SIP WEST)&lt;br /&gt;
&lt;br /&gt;
52.202.13.217   (Media EAST)&lt;br /&gt;
&lt;br /&gt;
44.236.207.157 (Media WEST)&lt;br /&gt;
|-&lt;br /&gt;
| Chat (Text Messaging)&lt;br /&gt;
| 5222&lt;br /&gt;
| XMPP&lt;br /&gt;
| TCP&lt;br /&gt;
| IP Address of PBX&lt;br /&gt;
|Not Assigned&lt;br /&gt;
|-&lt;br /&gt;
| Branch Office&lt;br /&gt;
| 4569&lt;br /&gt;
| IAX2&lt;br /&gt;
| UDP + TCP&lt;br /&gt;
| IP Address of PBX&lt;br /&gt;
|Not Assigned&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Note that when you are configuring a port forward in your router, that many routers let you specify an external and internal port.&amp;amp;nbsp; The internal port numbers that are forwarded to the PBX must match the numbers fromt the table above.&amp;amp;nbsp; However, for SSH, HTTP, or HTTPS you may use any external port number that is available in your router.'''&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPitomy_Communicator&amp;diff=5174</id>
		<title>IPitomy Communicator</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPitomy_Communicator&amp;diff=5174"/>
		<updated>2025-08-06T13:37:14Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Ipitomy Softphone */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Note: This Guide will assist the user in Installing and Using the IPitomy Communicator Softphone.&lt;br /&gt;
&lt;br /&gt;
We have two different soft phone installs. The “demo” version does not include the user list which shows all of the presence of all of the users on the system. This is intended for trial versions or for users who are on a shared instance of IPitomy in the Cloud. For obvious reasons, on a shared instance, it would not be good to display the presence of all the other users!&lt;br /&gt;
&lt;br /&gt;
The full commercial version is called IPitomy Communicator.&lt;br /&gt;
&lt;br /&gt;
This requires an available IPitomy Extension License as well as an IPitomy Soft Phone License. This does not have auto configuration, but is super simple to set up. When installing it, select IP620 as the device type for the extension so it will use an IPitomy license instead of an open license.&lt;br /&gt;
&lt;br /&gt;
This also requires the generation of a single API Key. If your PBX already has one generated for the IPitomy Communicator, you don't need to do anything else. If not, simply navigate to Applications=&amp;gt;API Key, name it Softphone [one word, capital S], and click Create.&amp;amp;nbsp; Be sure to enable it by checking the box labeled &amp;quot;Key Enable&amp;quot;, then save and apply changes.&lt;br /&gt;
&lt;br /&gt;
Links for SoftPhone:&lt;br /&gt;
[[File:Screenshot for communicator.png|alt=Screenshot for communicator|thumb|518x518px|Screenshot for communicator]]&lt;br /&gt;
[[File:Screenshot for communicator 2.png|thumb|519x519px|Screenshot for communicator 2]]&lt;br /&gt;
Demo version: [http://relay.ipitomy.com/softcall_demo/publish.htm http://relay.ipitomy.com/softcall_demo/publish.htm]&lt;br /&gt;
&lt;br /&gt;
Full version – IPitomy Communicator: [https://www.dropbox.com/s/6uhj12mwh8k7diu/CommunicatorSetup.zip?dl=0 &amp;lt;nowiki&amp;gt;[Here]&amp;lt;/nowiki&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
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&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
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&lt;br /&gt;
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&lt;br /&gt;
'''''&amp;lt;big&amp;gt;&amp;lt;u&amp;gt;IMPORTANT&amp;lt;/u&amp;gt;&amp;lt;/big&amp;gt;'''''&lt;br /&gt;
&lt;br /&gt;
To avoid an error, you must also install ''Visual C++ Redistributable for Visual Studio 2012 Update 4''.&lt;br /&gt;
&lt;br /&gt;
The file is here:  '''''&amp;lt;big&amp;gt;https://www.microsoft.com/en-us/download/details.aspx?id=30679&amp;lt;/big&amp;gt;'''''&lt;br /&gt;
&lt;br /&gt;
Scroll down just a little bit, click the download button, then select the x86 version (yes, even if you're running 64 bit Windows).&lt;br /&gt;
&lt;br /&gt;
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&lt;br /&gt;
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&lt;br /&gt;
&lt;br /&gt;
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&lt;br /&gt;
== '''Ipitomy Softphone'''&amp;lt;br/&amp;gt; ==&lt;br /&gt;
&lt;br /&gt;
'''Installation'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Installation Video'''&lt;br /&gt;
&lt;br /&gt;
[https://www.youtube.com/watch?v=p5gwURMp4eo &amp;lt;b&amp;gt;https://www.youtube.com/watch?v=p5gwURMp4eo&amp;lt;/b&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Park Button Video'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[https://www.youtube.com/watch?v=dPCVXiR6kf8 &amp;lt;b&amp;gt;https://www.youtube.com/watch?v=dPCVXiR6kf8&amp;lt;/b&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Install the Softphone, Follow the URL above and click the Installer link. Installation will begin at that time. You should have adminstrative rights to the PC you are installing to prior to installation. Accept the default location for install unless you have some other directory you wish to install to.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Setup'''&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Once isntalled open the program by clicking the program Icon. This should now be located on your Desktop as well as in the Program directory. The Program will open bringing up the user interface screen.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone3.jpg|File:SoftPhone3.jpg]][[File:SoftPhone4.jpg|File:SoftPhone4.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Add an account Press the Settings Icon which is the little Gear Icon next to the Ipitomy Logo Bar. Select Sip Accounts and Add an Account&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone5.jpg|File:SoftPhone5.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Select the Add Button and Enter the Following:&lt;br /&gt;
&lt;br /&gt;
Display Name = Name of user&lt;br /&gt;
&lt;br /&gt;
User Name = Extension Number&lt;br /&gt;
&lt;br /&gt;
User Password = Voicemail Password&lt;br /&gt;
&lt;br /&gt;
SIP Password = SIP Password for Extension&lt;br /&gt;
&lt;br /&gt;
SIP Server = IP of PBX (Local IP is OK for a Local Extension only, however for a remote phone this would need to the Public IP of the Network the PBX resides upon)&lt;br /&gt;
&lt;br /&gt;
Integration Port = Leave blank and the software uses 80 to verify its license with the PBX.&amp;amp;nbsp; If the softphone is remote and an external port other than 80 is forwarded to the PBX, enter that as the Integration Port (eg, 8080 external forwarded to 80 internal)&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
All other information can be left at Default.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Communicator-SIP Acc.jpg|File:Communicator-SIP Acc.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Be Sure to Click Enabled when you are returned to the account screen on the account you just created so it is made the live account&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone8.jpg|File:SoftPhone8.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Click close and you should be returned to the User interface and your account should register to the PBX.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone9.jpg|File:SoftPhone9.jpg]][[File:SoftPhone10.jpg|File:SoftPhone10.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Your Softphone is now ready to be used.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Import Extensions from the PBX to Contacts.'''&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Import all Extension in the PBX into the Contact list select User List - Manage Users - Ipitomy and then Refresh&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone11.jpg|File:SoftPhone11.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Select the PBX from the Host list&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
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[[File:SoftPhone14.jpg|File:SoftPhone14.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Click OK. The Softphone will now pull down an Extension list from the PBX which you configured.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone13.jpg|File:SoftPhone13.jpg]]&lt;br /&gt;
&lt;br /&gt;
== Troubleshooting Communictor errors ==&lt;br /&gt;
If you see the below error, here is the fix.&lt;br /&gt;
&lt;br /&gt;
They need to install vcredist.x86.  &amp;lt;nowiki&amp;gt;https://www.microsoft.com/en-us/download/details.aspx?id=30679&amp;lt;/nowiki&amp;gt; Make sure to choose the 32 bit ( regardless of their pc) called VSU_4\vcredist_x86.exe.  Either that or the dll file is actually missing.  Which is unlikely.&lt;br /&gt;
[[File:CommunicatorError1.jpg|left|thumb|753x753px]]&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=File:CommunicatorError1.jpg&amp;diff=5173</id>
		<title>File:CommunicatorError1.jpg</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=File:CommunicatorError1.jpg&amp;diff=5173"/>
		<updated>2025-08-06T13:35:10Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;jw&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=USB_Install_Dongle&amp;diff=5172</id>
		<title>USB Install Dongle</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=USB_Install_Dongle&amp;diff=5172"/>
		<updated>2025-07-30T16:22:33Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Creating a USB Install Dongle */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;If you are advised by Tech Support to perform a field install of PBX software, these are the instructions to create the USB dongle and how to use that to install the software on the PBX. Once completed, you will most likely need to upgrade the PBX to current software version via the normal process.&lt;br /&gt;
&lt;br /&gt;
== Creating a USB Install Dongle ==&lt;br /&gt;
&lt;br /&gt;
You'll find both the LiveUSBCreator.zip file here: [https://www.dropbox.com/sh/hsdsk0lr00rnlr5/AABTL5y38oBfu3vKvBhrodGba/USB_Disk/install_disk?dl=0&amp;amp;preview=LiveUSBCreator.zip Live USB Creator] The version 6 install ISO can be found here: [https://www.dropbox.com/sh/hsdsk0lr00rnlr5/AADLN4zNL0MSNy7SAjfXhq12a/IPitomy/IP1100%2B?dl=0&amp;amp;preview=IPitomyPBXReinstall.iso Version 6 Install Image]&amp;amp;nbsp;(NOTE: Installing version 6 REQUIRES a 64-bit compatible system, please check with tech support if you are unsure about your pbx).&lt;br /&gt;
&lt;br /&gt;
When you click on the links, they will take you to a Dropbox.  There you will navigate to the right side of the screen and click on the download button to download the files.&lt;br /&gt;
&lt;br /&gt;
[[File:USB999.png|frameless|600x600px]]&lt;br /&gt;
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&lt;br /&gt;
[[File:USB25555.png|frameless|600x600px]]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br /&amp;gt;Download the file, LiveUSBCreator.zip and extract it's contents. This will create a folder called LiveUSBCreator Download the file, IPitomyPBXReinstall.iso. Put this file in the newly created LiveUSBCreator folder Put a formated (FAT32), empty 1GB USB drive in the system.  Enter the folder LiveUSBCreator.  Right click on file: liveusb-creator application, and Run as Administrator.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Usb2.png|frame|alt=|none]]&lt;br /&gt;
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[[File:USB.png|frame|alt=|none]]&lt;br /&gt;
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&amp;lt;br/&amp;gt;[[File:001.jpg|File:001.jpg]]&lt;br /&gt;
&lt;br /&gt;
The USB drive letter should be visible in the &amp;quot;Target Device&amp;quot; box. If not, then select it from the list. Click the [Browse] button under &amp;quot;Use existing Live CD&amp;quot; Browse to and select &amp;quot; Install .iso&amp;quot; Click [Open]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:002.jpg|File:002.jpg]]&lt;br /&gt;
&lt;br /&gt;
Click [Create Live USB]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:003.jpg|File:003.jpg]]&lt;br /&gt;
&lt;br /&gt;
After a few minutes, the Install USB will have been created:&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:004.jpg|File:004.jpg]]&lt;br /&gt;
&lt;br /&gt;
Click the [X] in the upper right corner to close the program and properly eject the USB drive from the system&lt;br /&gt;
&lt;br /&gt;
== Using the USB Install Dongle&amp;lt;br/&amp;gt; ==&lt;br /&gt;
&lt;br /&gt;
[[File:Tech Bulletin 2011-009 - IPitomy ERD v 2.0.pdf|File:Tech Bulletin 2011-009 - IPitomy ERD v 2.0.pdf]]&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=File:USB25555.png&amp;diff=5171</id>
		<title>File:USB25555.png</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=File:USB25555.png&amp;diff=5171"/>
		<updated>2025-07-30T16:22:10Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;jw&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=File:USB999.png&amp;diff=5170</id>
		<title>File:USB999.png</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=File:USB999.png&amp;diff=5170"/>
		<updated>2025-07-30T16:20:26Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;jw&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPPBX_Manual_System_Admin_Adding_and_Deleting_Extensions&amp;diff=5169</id>
		<title>IPPBX Manual System Admin Adding and Deleting Extensions</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPPBX_Manual_System_Admin_Adding_and_Deleting_Extensions&amp;diff=5169"/>
		<updated>2025-05-21T12:46:07Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: jw&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;[[File:Crap.png|left|frameless]]&lt;br /&gt;
'''Extensions'''&lt;br /&gt;
&lt;br /&gt;
Extensions define where specific people or departments can be reached in an organization. They should be setup first in the system. The following is a list of the various settings/parameters that will can be updated for each extension. The parameters you configure for the extensions will vary based on the customer’s general business practices.&lt;br /&gt;
&lt;br /&gt;
*General Settings&lt;br /&gt;
*Forwarding Settings&lt;br /&gt;
*Advanced Network Settings&lt;br /&gt;
*Advanced Voicemail Settings&lt;br /&gt;
*Advanced Allow Codecs Settings&lt;br /&gt;
*Advanced Calling Permissions&lt;br /&gt;
*Advanced Follow-Me&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
=== '''Add/Import Tab'''&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
The '''Add/Import '''tab allows you to create new extensions or edit existing extensions.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;Use the Setup Worksheet to upload the CSV file and automatically create the Extension information. Please refer to the Setup Worksheet section of this guide for details.&lt;br /&gt;
&lt;br /&gt;
[[File:Addimporttab.png|center|Addimporttab.png]] [[File:Createextension.png|center|Createextension.png]]&lt;br /&gt;
&lt;br /&gt;
{| class=&amp;quot;wikitable&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| '''Sections/Fields'''&lt;br /&gt;
| &amp;lt;center&amp;gt;'''Description'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| '''Extension Name'''&lt;br /&gt;
| This is the name of the person that the extension will be assigned to (using this device).&lt;br /&gt;
|-&lt;br /&gt;
| '''Email Address'''&lt;br /&gt;
| This is the user’s email address (optional)&lt;br /&gt;
|-&lt;br /&gt;
| '''Ext. #'''&lt;br /&gt;
| The extension number that is assigned to this device.&lt;br /&gt;
|-&lt;br /&gt;
| '''Device Type'''&lt;br /&gt;
| This is the type of device that will be using this extension (i.e. IP550).&lt;br /&gt;
|-&lt;br /&gt;
| '''MAC'''&lt;br /&gt;
| This is the MAC ID of the device (optional).&lt;br /&gt;
|-&lt;br /&gt;
| '''AutoNumber/Start At'''&lt;br /&gt;
| Selecting this option allows you to automatically number the extensions that you need to add. To use this feature, simply enter the extension number you want to start with then select (place a checkmark) the AutoNumber option.&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
''Table 15Create Extension Fields and Descriptions''&lt;br /&gt;
&lt;br /&gt;
==== Add/Create Extensions&amp;lt;br/&amp;gt; ====&lt;br /&gt;
&lt;br /&gt;
This section describes in detail how to create a new extension.&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From any page of the PBX Admin system, click on the button next to the '''Destinations''' link to expand the menu. The '''Destination''' menu opens and displays the options available.&lt;br /&gt;
#Click on the '''Extensions''' link. The Extensions page opens and displays a listing of extensions (if ones already exists).&lt;br /&gt;
#Click on the box to the left of the '''Extensions '''field. This value defaults to “'''10'''”. Enter the number of extensions you want to add and then click the '''ADD''' button.&lt;br /&gt;
#The '''Create Extensions''' page appears displaying the number of rows that was specified. Enter information for the extension in these fields. See table above for details.&lt;br /&gt;
#:You can have the system automatically number the extensions you want to create by clicking the '''AutoNumber''' checkbox located on the top left hand corner of the '''Create Extensions''' page.&lt;br /&gt;
#Click on the '''CREATE''' button when all the extension information you want to create is entered. The system responds with a message indicating the results of adding the new extension(s). You should see a “'''SUCCESS'''” message.&lt;br /&gt;
#If there is an error, you will see an “'''ERROR'''” indicated under the Results field. An error is typically due to an extension number that is being duplicated (already existing in the system). Make the necessary adjustments to correct the error then click the '''CREATE''' button.&lt;br /&gt;
#Click on the [[File:Savechanges.png]] button, to save the changes.&lt;br /&gt;
#Select the '''Apply Changes''' link located on the right hand corner of the page, to commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
=== '''Search Tab'''&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
This tab allows the user to search the extensions for keywords in the fields: Class of Service, Departments, Name, Numbers, CID Names, CID Numbers, Email, or Status .This section describes in detail how to search for existing extensions. [[File:Searchextensions.png|center|Searchextensions.png]]&amp;lt;br/&amp;gt;The following table describes the types of search parameters the system will perform.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{| class=&amp;quot;wikitable&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| '''Sections/Fields'''&lt;br /&gt;
| &amp;lt;center&amp;gt;'''Description'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| '''Class of Service'''&lt;br /&gt;
| This is the class of service that is assigned to the extension(s).&lt;br /&gt;
|-&lt;br /&gt;
| '''Departments'''&lt;br /&gt;
| This is the department that is assigned to the extension(s).&lt;br /&gt;
|-&lt;br /&gt;
| '''Names'''&lt;br /&gt;
| This is the name of the user assigned to the extension(s).&lt;br /&gt;
|-&lt;br /&gt;
| '''Numbers'''&lt;br /&gt;
| This is the extension number.&lt;br /&gt;
|-&lt;br /&gt;
| '''CID Names'''&lt;br /&gt;
| The caller ID name that is associated with the extension(s).&lt;br /&gt;
|-&lt;br /&gt;
| '''CID Numbers'''&lt;br /&gt;
| This is the caller ID number that is associated with the extension(s).&lt;br /&gt;
|-&lt;br /&gt;
| '''Emails'''&lt;br /&gt;
| Email address that is associated with the extension(s).&lt;br /&gt;
|-&lt;br /&gt;
| '''Status'''&lt;br /&gt;
| This is the status of the extension(s).&lt;br /&gt;
|-&lt;br /&gt;
| '''Match Search Filter'''&lt;br /&gt;
| Exactindicates that you want the search to match exactly as the search criteria that is entered.&lt;br /&gt;
Partialindicates that you want to partially match the search criteria entered.&lt;br /&gt;
&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
==== Search Extension&amp;lt;br/&amp;gt; ====&lt;br /&gt;
&lt;br /&gt;
This section describes how to search for extensions.&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''Extensions''' page, click the '''Search Tab'''. The list of extensions that are in the system appears with search options at the top section of the page.&lt;br /&gt;
#Click the [[File:Dropdownlist.png]] drop-down arrow icon next to the '''Class of Service''' list.&lt;br /&gt;
&lt;br /&gt;
Select the desired '''search criteria''' then enter the parameters in the box to the right of the list then click the '''Search''' button. If the system finds any extensions matching your search parameter, it will display the information in the extensions window.&lt;br /&gt;
&lt;br /&gt;
If the field is left blank, the system will bring all the extensions.&lt;br /&gt;
&lt;br /&gt;
=== View Tab (Extensions)&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
This tab allows the user to sort the display of extensions by Phone Model. Once sorted, phone key settings can be mass edited for phones of the same model. This section describes in detail how to view existing extensions.&lt;br /&gt;
&lt;br /&gt;
[[File:Extensionsviewpage.png|center|Extensionsviewpage.png]]&lt;br /&gt;
&lt;br /&gt;
==== Edit or View Extension&amp;lt;br/&amp;gt; ====&lt;br /&gt;
&lt;br /&gt;
This section describes in detail how to view or edit extension details.&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#Navigate to the '''Destinations'''=&amp;gt;'''Extensions''' page.&lt;br /&gt;
#Select the [[File:Penciledit.png]] or the [[File:Pencilextensionedit.png]] icon to the right of the extension name you want to view or edit. The pencil edits the PBX settings and the pencil with the handset behind it edits the Phone settings.&lt;br /&gt;
#The '''Edit Extensions''' page displays with setting details for the extension.&lt;br /&gt;
#Make the necessary changes to the extension.&lt;br /&gt;
#Click on the [[File:Savechanges.png]] button to save the changes.&lt;br /&gt;
#Click the '''Apply Changes''' link located at the right hand corner of the top of the page, to commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
You can edit multiple extensions by selecting (placing a checkmark in the box next to extension name). Only the fields being changed (that is common for all extensions selected) will be modifiedi.e. Status or call group, etc.).&lt;br /&gt;
&lt;br /&gt;
Another shortcut that the system provides you is the '''Previous''' and '''Next''' button located on the top left corner of the '''Edit Extension''' page. Use these buttons to navigate backward or forward to find the extensions you want to view or modify.&lt;br /&gt;
&lt;br /&gt;
==== Mass Edit PBX Extension Settings&amp;lt;br/&amp;gt; ====&lt;br /&gt;
&lt;br /&gt;
This section describes in detail how to view or edit extension details.&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''Destinations'''=&amp;gt;'''Extensions''' page, click on the '''View''' tab. A list of extensions appears.&lt;br /&gt;
&lt;br /&gt;
You can sort the list by phone model by selecting (placing a checkmark) the'''Phone Model &amp;amp; Settings''' option located in the top left hand corner of the screen. If you are using this method, you can also Mass Edit Phone Key settings. See the section below for steps.&lt;br /&gt;
&lt;br /&gt;
#Select (place a checkmark) in the box next to the extension name(s) you want to view or edit.&lt;br /&gt;
#Click on the '''Edit PBX Settings''' button. The extension details page appears. On the top left corner of the screen, you will see the extension numbers that you are viewing or editing.&lt;br /&gt;
#Make the necessary changes to the extension settings then click on the [[File:Savechanges.png]] button to save the changes.&lt;br /&gt;
#Click the '''Apply Changes''' link located at the right hand corner of the top of the page, to commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
==== Mass Edit Phone Key Settings&amp;lt;br/&amp;gt; ====&lt;br /&gt;
&lt;br /&gt;
This section of the Administration Guide describes how to mass edit the key settings for extensions using the same phone model. [[File:Masseditfeature.png|center|Masseditfeature.png]]&amp;lt;br/&amp;gt;'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''Destinations'''-&amp;gt;'''Extensions''' page, click on the '''View '''tab.&lt;br /&gt;
#Select the '''Phone Models and Settings''' option at the top left hand corner of the list.&lt;br /&gt;
#The system will display a list of the extensions in a grouping of phone types. Select the group of phones you want to edit. A listing of all the extensions in the phone group will appear.&lt;br /&gt;
#Select the box next to the name of the extensions you want to update or click on box next to the '''Name''' field at the top of the column to select all the extensions.&lt;br /&gt;
#Click the '''Edit Phone Settings''' button. The system will take you to the '''Key Settings''' page for the phones. At the top of the page you will see a list of all the extensions that you are updating.&lt;br /&gt;
#By default, the '''Only Save Changed Fields''' option Is selected, de-select if needed.&lt;br /&gt;
#Once all the changes are made, select the '''Save and Restart''' '''Phone''' button to save changes and reboot the phones so they can pull down their updated configuration file.&lt;br /&gt;
&lt;br /&gt;
==== Delete Extension&amp;lt;br/&amp;gt; ====&lt;br /&gt;
&lt;br /&gt;
This section describes in detail how to delete existing extensions.&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#Navigate to the '''Destinations'''-&amp;gt;'''Extensions''' page&lt;br /&gt;
#Select the [[File:Deleteselected.png]] icon to the right of the extension name you want to Delete&lt;br /&gt;
#The extension is deleted and a confirmation message will appear. Click '''OK''' on the message window.&lt;br /&gt;
#The system returns you to the '''Extensions''' page. The extension that was just deleted will no longer appear on the list of extensions.&lt;br /&gt;
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
==== Delete Multiple Extension&amp;lt;br/&amp;gt; ====&lt;br /&gt;
&lt;br /&gt;
This section describes in detail how to delete multiple extensions.&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#Navigate to the '''Destinations'''=&amp;gt;'''Extensions''' page&lt;br /&gt;
#Select (place a checkmark) in the boxes next to the extension name(s) you want to delete.&lt;br /&gt;
#Click on the Delete All button. The selected extension will be removed from the list and deleted from the database.&lt;br /&gt;
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
=== Extensions - General Settings Section&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
Once the extensions have been added to the database, you can edit the settings for each of the extensions.&lt;br /&gt;
&lt;br /&gt;
[[File:Extensionsgeneralsettings.png|center|Extensionsgeneralsettings.png]]&lt;br /&gt;
&lt;br /&gt;
{| class=&amp;quot;wikitable&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| '''Sections/Fields'''&lt;br /&gt;
| &amp;lt;center&amp;gt;'''Description'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''Name'''&lt;br /&gt;
| Name of the user associated with the extensions being created&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''Number'''&lt;br /&gt;
| Extension number for this person or department. This must be 3 to 4 digits in length&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''Email'''&lt;br /&gt;
| Email address for the person assigned to the extension. This will allow the system to forward email messages to the address of the person at the extension when properly configured.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''Status'''&lt;br /&gt;
| Active = currently in use&lt;br /&gt;
Disabled = currently not in use&lt;br /&gt;
&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''Class of Service'''&lt;br /&gt;
| This is the service type for the extension. When initially created, the PBX will set this to the COS you have definined as the system default class of service on the PBX SetupGeneral Page&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''PIN'''&lt;br /&gt;
| This is the number used to access the extensions voicemail and can be between 1 and 6 digits long. The default setting is for the PIN to be the extension number. Be sure to instruct users to change the PIN to avoid unauthorized use.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''Ring Time'''&lt;br /&gt;
| This is the time in seconds that a call will ring before it is considered unanswered. Ring time must be between 1 and 100 seconds in length.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''Call Group'''&lt;br /&gt;
| This number assigns this extension to a group with a similar purpose (e.g., Sales or Customer Service). Multiple call groups can be assigned to each extension by putting a comma between the group numbers. The call groups also define which Pickup Groups can answer calls to this extension.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''Pickup Group'''&lt;br /&gt;
| This number should match any Call Group number entered on an extension. It defines the Call Group Numbers this extension can pickup remotely by pressing 99.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''Apply Schedule'''&lt;br /&gt;
| When an extension is created, a schedule destination is created automatically. This schedule is not activated until the Apply Schedule box is selected. When it is selected, all calls sent directly to this extension must first pass through the extension’s schedule and will be routed accordingly. Extension schedules will appear with the name of the extension (e.g., Extension 123 would appear as “ext_123”). (See the Schedules section of this guide for more information.)&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''CID Override'''&lt;br /&gt;
| If enabled, the user will be able to override the Caller ID settings. When the user places a call, the original assigned CID will be bypassed and the name and number will that is entered in the CID Name and Number fields will be sent instead. Always check with your provider that CID override is allowed before configuring.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''CID Name'''&lt;br /&gt;
| If the CID Override parameter is enabled, this is the Caller ID name that will be seen by the recipient when an outbound call is placed. Always check with your provider that CID override is allowed before configuring.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''CID Number'''&lt;br /&gt;
| If the CID Override parameter is enabled, this is the Caller ID number that will be seen by the recipient when an outbound call is placed. Always check with your provider that CID override is allowed before configuring.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''Detect Fax'''&lt;br /&gt;
| If checked, calls from an Analog or T1/PRI card that route direct to this extension will spend a period of time (Defined under PBX SetupGeneral) checking if the call is a fax. During this time, the PBX holds the call; if fax tone is detected, the call will be passed along to the destination defined for Route Fax To, otherwise it will pass the call to the extension after the detection time has expired.  Note: Fax detect does not work on inbound calls from a SIP trunk.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''Route Fax To'''&lt;br /&gt;
| Define where calls will be routed if fax tone is detected.&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
''Table 17General Extension Settings and Descriptions''&lt;br /&gt;
&lt;br /&gt;
=== Extensions - Forward Settings Section&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
The extensions forwarding settings are made to be very user friendly. The settings may be modified from the Smart Personal Console, changed from your telephone extension, or changed remotely from any telephone (including cell phones), using the touch-tone key pad. [[File:Extensionsforwardsettings.png|center|Extensionsforwardsettings.png]]&amp;lt;br/&amp;gt;Forward settings routes calls to a different destination. These settings can be:&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{| class=&amp;quot;wikitable&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| '''Sections/Fields'''&lt;br /&gt;
| &amp;lt;center&amp;gt;'''Description'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''Unconditional'''&lt;br /&gt;
| Always route calls to a specific destination.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''Busy'''&lt;br /&gt;
| Route calls to a specific destination when the extension is Paused, or when the user Rejects/Ignores the call via softkey.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''No Answer'''&lt;br /&gt;
| Route calls to a specific destination when a call is not answered in the defined Ring Time&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;text-align: center&amp;quot; | '''Unavailable'''&lt;br /&gt;
| Route calls to a specific destination when a phone is turned off, is not registered with the system, or has reached its call limit (as set in the IPitomy IP PBX).&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
''Table 18Extension Forward Settings and Descriptions''&lt;br /&gt;
&lt;br /&gt;
==== Enable/Disable Forward Settings ====&lt;br /&gt;
&lt;br /&gt;
The following outlines steps to enable or disable forward settings:&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#Pick the setting to be modifiedUnconditional, Busy, No Answer or Unavailable.&lt;br /&gt;
#Select '''Enabled''' or '''Disabled'''. Disabled turns the forward setting off. Enabled turns the forward setting on.&lt;br /&gt;
#Select either Phone Number or Destination. Phone Number allows you to enter the digits you want dialed, like a PSTN number. Destination will bring up the standard dropdown list of destinations in the system; Extensions, Groups, etc.&lt;br /&gt;
#Enter the Phone Number or select the Destination you would like the PBX to route to when meeting the forwarding requirements.&lt;br /&gt;
#Click the [[File:Savechanges.png]] button to save the changes. The system returns you to the '''Edit Extensions''' page.&lt;br /&gt;
#Click the '''Apply Changes''' link located on the right hand corner of the page, to commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
==== Change Unconditional Forwarding via Keypad ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
Only unconditional forwarding can be changed from a touch-tone keypad. Enter the following code to set the unconditional forwarding setting.&lt;br /&gt;
&lt;br /&gt;
#Dial '''&amp;lt;nowiki&amp;gt;*90&amp;lt;/nowiki&amp;gt;''' to '''disable''' forwarding.&lt;br /&gt;
#Dial '''&amp;lt;nowiki&amp;gt;*91&amp;lt;/nowiki&amp;gt;''' to '''enable''' forwarding.&lt;br /&gt;
#Dial '''&amp;lt;nowiki&amp;gt;*92&amp;lt;/nowiki&amp;gt;''' to '''set''' the forwarding number.&lt;br /&gt;
&lt;br /&gt;
==== Change Unconditional Forwarding via PC ====&lt;br /&gt;
&lt;br /&gt;
The following outlines the steps for the end user to change the extension forwarding setting from a PC using the Smart Personal Console. The administrator will need to have enabled Allow User to control Forwarding under the extensions calling permissions.&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#Browse the internet to the '''Smart Personal Console''' page.&lt;br /&gt;
#Login.&lt;br /&gt;
#Click the '''My Account''' link to access the forwarding page.&lt;br /&gt;
#Enable/Disable the desired forwarding setting.&lt;br /&gt;
#Select either Phone Number or Destination. Phone Number allows you to enter the digits you want dialed, like a PSTN number. Destination will bring up the standard dropdown list of destinations in the system; Extensions, Groups, etc.&lt;br /&gt;
#Enter the Phone Number or select the Destination you would like the PBX to route to when meeting the forwarding requirements.&lt;br /&gt;
#Click the [[File:Savechanges.png]] button to save the changes&lt;br /&gt;
&lt;br /&gt;
==== Change Forwarding Number While Away from an Extension ====&lt;br /&gt;
&lt;br /&gt;
Only unconditional forwarding can be changed from a touch-tone keypad.&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#Call into the '''Automated Attendant''' (menu).&lt;br /&gt;
#Select the touch-tone digit that routes to the Forwarding Gateway.&lt;br /&gt;
#The system will prompt for an '''Extension Number''' and '''Password'''.&lt;br /&gt;
#The system will indicate if extension forwarding is '''Enabled''' or '''Disabled'''.&lt;br /&gt;
#Pressing “'''1'''” toggles between Enabled and Disabled.&lt;br /&gt;
#Pressing “'''2'''” allows the forwarding destination to be modified.&lt;br /&gt;
&lt;br /&gt;
=== Extensions - Advanced Settings ===&lt;br /&gt;
&lt;br /&gt;
==== Extensions - Network Settings Section ====&lt;br /&gt;
&lt;br /&gt;
Network settings automatically register in the extension through the system. These settings represent registration and identification information. The system (extension) defaults should not be changed without advanced knowledge of the behaviors of the particular settings.&lt;br /&gt;
&lt;br /&gt;
[[File:Extensionsadvsettings.png|center|Extensionsadvsettings.png]]&lt;br /&gt;
&lt;br /&gt;
{| class=&amp;quot;wikitable&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| '''Sections/Fields'''&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | &amp;lt;center&amp;gt;'''Description'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| '''SIP Password'''&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | Password for the user to access IP PBX web-based administration system. Use a combination of uppercase letters, lowercase letters, and numbers.&lt;br /&gt;
|-&lt;br /&gt;
| '''Generate'''&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | If clicked, the system will automatically generate a password for the extension.&lt;br /&gt;
|-&lt;br /&gt;
| '''Password Strength'''&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | This color code bar indicates the strength of the password being assigned for the extension. The strengths are represented with the following colors:&lt;br /&gt;
[[File:Pwrodstrength.png]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
|-&lt;br /&gt;
| '''Location'''&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | This allows the user to tell the PBX whether to expect this extension to register as a local extension (LAN) or as a remote extension (WAN).&lt;br /&gt;
|-&lt;br /&gt;
| '''NAT'''&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | &lt;br /&gt;
{|&lt;br /&gt;
|-&lt;br /&gt;
| &lt;br /&gt;
| IMPORTANT: This setting should be ENABLED (checked) unless otherwise instructed. Please contact an IPitomy’s Technical Support Group for assistance or more information.&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
|-&lt;br /&gt;
| '''HOST'''&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | &lt;br /&gt;
{|&lt;br /&gt;
|-&lt;br /&gt;
| &lt;br /&gt;
| IMPORTANT: This should be set to DYNAMIC unless otherwise instructed. Please contact IPitomy’s Technical Support Group for assistance.&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
|-&lt;br /&gt;
| '''Phone Type'''&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | The phone type is a drop down list for selecting which IP phone hardware is being used on the extension. IPitomy supports Aastra phones as well as our own IP550 and IP120 phones, and will be adding additional phone types in the future. When the phone type is selected, another configuration option is available to program the button mapping of each telephone model. The IPitomy IP PBX supports a variety of pre-programmed buttons like BLF, park, voicemail, as well as custom configurable speed dial buttons. Each phone can be configured for its own unique set of buttons.&lt;br /&gt;
|-&lt;br /&gt;
| '''Phone MAC'''&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | All of the IP phones have a MAC Address. The MAC ID identifies the piece of equipment for configuration. The auto configuration features of IPitomy rely on the MAC address to load the proper configuration files into the telephone when changes are made in the Web-based interface. The configuration files are stored on the IPitomy IP PBX and used when the phone powers back on after a power down cycle. If the configuration files have been updated when the phone powers back on, a new configuration is loaded into the phone. When the new configuration file is loaded, manual settings on the phone take priority and will be kept intact during the upgrade. Note that at this time only Aastra phones require and utilize the MAC address in the phone settings.&lt;br /&gt;
|-&lt;br /&gt;
| '''Qualify'''&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | This is the interval of time between checking registration for the phone. Default is set to 8000 and should not be changed unless instructed by an IPitomy representative.&lt;br /&gt;
|-&lt;br /&gt;
| '''DTMF Mode'''&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | This is where you set what kind of DTMF signaling the extension will use. The dropdown lists options are:&lt;br /&gt;
*'''rfc2833''' (recommended in the HD Phone models)&lt;br /&gt;
*'''auto'''&lt;br /&gt;
*'''info''' (recommended in the IPitomy legacy phone models IP120 &amp;amp; IP550)&lt;br /&gt;
*'''inband'''&lt;br /&gt;
&lt;br /&gt;
|-&lt;br /&gt;
| '''User Type'''&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | This is defaulted to Friend and should not be changed unless instructed to do so by an IPitomy representative.&lt;br /&gt;
|-&lt;br /&gt;
| '''Call Limit'''&lt;br /&gt;
| This is the number of concurrent calls allowable at an extension. The Call Limit selected must be between 1 and 99. Default is set to 4. It needs to be above 0 for BLF keys to function.&lt;br /&gt;
| &lt;br /&gt;
|-&lt;br /&gt;
| '''Can Reinvite'''&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | This parameter allows a device to reconnect calls midstream.&lt;br /&gt;
*YES = if the phone type allows the re-invite feature&lt;br /&gt;
*NO = if the phone type does not allow the re-invite feature&lt;br /&gt;
*N/A = accepts the system wide default defined in the System Setup section of the Administration System. If the default setting is acceptable and works within your business, we recommend leaving the parameter set to N/A.&lt;br /&gt;
&lt;br /&gt;
|-&lt;br /&gt;
| '''RTP Keep-alive'''&lt;br /&gt;
| Number of seconds, when a RTP Keepalive packet will be sent if no other RTP traffic on that connection. Default 0 (no RTP Keepalive)&lt;br /&gt;
|-&lt;br /&gt;
| '''Send Remote Party ID'''&lt;br /&gt;
|  yes|no : If a Remote-Party-ID SIP header should be sent. Default no&lt;br /&gt;
|-&lt;br /&gt;
| '''Trust Remote Party ID'''&lt;br /&gt;
| yes|no : If Remote-Party-ID SIP header should be trusted. Default no&lt;br /&gt;
|-&lt;br /&gt;
| '''Insecure'''&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | This parameter allows you to specify how to handle connections with peers. Explanation of the different options available on the drop-down list are:&lt;br /&gt;
*'''PORT''' = Ignore the port number where authentication came from.&lt;br /&gt;
*'''INVITE''' = Do not require the initial invite to authenticate.&lt;br /&gt;
*'''PORT+INVITE''' = Do not require initial invite to authenticate and ignore the port where the request came from.&lt;br /&gt;
*'''YES''': To match a peer based by IP Address only and not the port.&lt;br /&gt;
*'''VERY''': To allow registered hosts to call without re-authenticating. This is the default setting.&lt;br /&gt;
&lt;br /&gt;
|-&lt;br /&gt;
| '''Music On Hold'''&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | This setting allows the user to select a different Music On Hold playlist for their extension then the system default playlist.&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
''Table 19Extensions Advanced Networking Settings and Descriptions''&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
===== Edit Extensions - Network Settings =====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''Edit Extensions''' page, locate the extension that you want to update. Click the [[File:Penciledit.png]] (edit extensions) icon to the right of the name.&lt;br /&gt;
You can also '''edit multiple''' extensions by selecting (placing a checkmark) in the boxes to the left of the extensions you want to update. Click the Edit PBX Settings button located at the top right hand corner of the list.You will see the extensions that are currently being updated. Make sure that the “'''Only save the changed field'''s” box is selected.&lt;br /&gt;
&lt;br /&gt;
#The '''Extension Details''' page appears. Select the link '''Advanced''' to open the '''Advanced Settings''' page.&lt;br /&gt;
#Make the necessary changes in the '''Network Settings''' section of the page.&lt;br /&gt;
#Click the [[File:Savechanges.png]] button to save the changes. The system returns you to the '''Edit Extensions''' page.&lt;br /&gt;
#Click the '''Apply Changes''' link located on the right hand corner of the page, to commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
==== Extensions - Voicemail Settings Section ====&lt;br /&gt;
&lt;br /&gt;
These settings manage voicemail messaging and routing. Mailboxes created when the extension is built can either be edited on the Advanded section of the extension or on the Voicemail section under Destinations. [[File:Extensionsvmail.png|center|Extensionsvmail.png]]&amp;lt;br/&amp;gt;This table describes the voicemail setting options for each extension.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Sections/Fields'''&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Description'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Mailbox'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | This is the number associated with the extension.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Attach to Email'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Select to enclose the message received in a notification e-mail as an attachment to the email address entered for the extension. An audio file (.Wav) will be the attachment. This requires for Unified Messaging to be configured on the PBX.&lt;br /&gt;
'''YES''' = attach message to email.&lt;br /&gt;
&lt;br /&gt;
'''NO''' =do not attach message to email.&lt;br /&gt;
&lt;br /&gt;
'''N/A''' = accepts the system wide default defined under PBX Setup=&amp;gt;Voicemail. If the default setting is acceptable and works within your business, we recommend leaving the parameter set to N/A.&lt;br /&gt;
&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Delete After Emailing'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Delete the voicemail after it has been emailed to the email address provided for the extension in General Settings.&lt;br /&gt;
'''YES''' = delete message after emailing.&lt;br /&gt;
&lt;br /&gt;
'''NO''' = do not delete message after emailing.&lt;br /&gt;
&lt;br /&gt;
'''N/A''' = accepts the system wide default defined under PBX Setup=&amp;gt;Voicemail. If the default setting is acceptable and works within your business, we recommend leaving the parameter set to N/A.&lt;br /&gt;
&lt;br /&gt;
'''Note: This option should not be enabled with turn old after emailing. If you enable both, the message will not be emailed but it will be deleted.'''&lt;br /&gt;
&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Turn Old After Emailing'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | After emailing, the system moves the voicemail message to the Old folder.&lt;br /&gt;
'''YES''' = will move message to Old messages folder after emailing.&lt;br /&gt;
&lt;br /&gt;
'''NO''' = messages will not be moved after emailing.&lt;br /&gt;
&lt;br /&gt;
'''Note: This option should not be enabled with delete after emailing. If you enable both, the message will not be emailed but it will be deleted.'''&lt;br /&gt;
&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Say Caller ID'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | State Caller ID prior to playback of the message.&lt;br /&gt;
'''YES''' = play caller id prior to message content.&lt;br /&gt;
&lt;br /&gt;
'''NO''' = do not play caller id prior to message content.&lt;br /&gt;
&lt;br /&gt;
'''N/A''' = accepts the system wide default defined under PBX Setup=&amp;gt;Voicemail. If the default setting is acceptable and works within your business, we recommend leaving the parameter set to N/A.&lt;br /&gt;
&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow Review'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow callers to review a message after they have recorded it.&lt;br /&gt;
'''YES''' = give callers leaving messages the option to review and rerecord the message they are leaving.&lt;br /&gt;
&lt;br /&gt;
'''NO''' = do not give callers the option to rerecord.&lt;br /&gt;
&lt;br /&gt;
'''N/A''' = accepts the system wide default defined under PBX Setup=&amp;gt;Voicemail. If the default setting is acceptable and works within your business, we recommend leaving the parameter set to N/A.&lt;br /&gt;
&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow Operator'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow pressing “0” during the voicemail greeting to reach the system-wide operator.&lt;br /&gt;
'''YES''' = allow dialing 0 from mailbox.&lt;br /&gt;
&lt;br /&gt;
'''NO''' = disallow dialing 0 from mailbox.&lt;br /&gt;
&lt;br /&gt;
'''N/A''' = accepts the system wide default defined under PBX Setup=&amp;gt;Voicemail. If the default setting is acceptable and works within your business, we recommend leaving the parameter set to N/A.&lt;br /&gt;
&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Play Envelope Message'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Play the time of call prior to the message.&lt;br /&gt;
'''YES''' = enabled.&lt;br /&gt;
&lt;br /&gt;
'''NO''' = disabled.&lt;br /&gt;
&lt;br /&gt;
'''N/A''' = accepts the system wide default defined under PBX Setup=&amp;gt;Voicemail. If the default setting is acceptable and works within your business, we recommend leaving the parameter set to N/A.&lt;br /&gt;
&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Auto Delete Voicemail In'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Define the number of days in which voicemail messages are to be automatically deleted from a mailbox. If this is set to “'''0'''” (zero) the voicemail message will never expire.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow Dial Out Access'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | This allows the ‘Dial Out’ feature when a user is listening to their voicemail.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow Mobile Web Access'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default is No.  Set to yes to allow users to use the Mobile App as well as their Smart Personal Console.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Exclude from Directory'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | This indicates whether to exclude this extension from the directory.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Mailbox Operator'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If a caller presses “'''0'''” (zero) while listening to your mailbox greeting, the caller will be routed to this destination. Also, this is where a user will be sent if he dials ”0” from this extension. '''Set this to None to use the system default''' '''which is set under PBX SetupGeneral.'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Mailbox Exit Destination'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | The Mailbox Exit Destination is where the system will route a caller who presses '''&amp;lt;nowiki&amp;gt;#&amp;lt;/nowiki&amp;gt;''' when they finish leaving a voicemail message. '''Setting this to None will use the system default, which is set under PBX SetupGeneral.'''&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
===== Edit Voicemail Settings =====&lt;br /&gt;
&lt;br /&gt;
The following outlines the steps to set voicemail parameters.&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''Edit Extensions''' page, locate the extension that you want to update. Click the [[File:Penciledit.png]] (edit extensions) icon to the right of the name.&lt;br /&gt;
You can also '''edit multiple''' extensions by selecting (placing a checkmark) in the boxes to the left of the extensions you want to update. Click the Edit PBX Settings button located at the top right hand corner of the list.You will see the extensions that are currently being updated. Make sure that the “'''Only save the changed field'''s” box is selected.&lt;br /&gt;
&lt;br /&gt;
#The '''Extension Detail '''page appears. Select the link to open the '''Advanced Settings''' page.&lt;br /&gt;
#Make the necessary changes in the '''Voicemail Settings''' section of the page.&lt;br /&gt;
#Click the [[File:Savechanges.png]] button to save the changes. The system returns you to the '''Edit Extensions''' page.&lt;br /&gt;
#Click the '''Apply Changes''' link located on the right hand corner of the page, to commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
==== Extensions - Allow CODECs Section ====&lt;br /&gt;
&lt;br /&gt;
These transmission speeds are configured by the service provider and designed to automatically register in the extension through the system. [[File:Codecssettings.png|center|Codecssettings.png]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Sections/Fields'''&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Description'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''CODEC Permissions (Allow CODECs)'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allows the administrator to define which codec the extension should use, and specify a priority from top down.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #000000;  border-bottom:0.0069in solid #000000;  border-left:0.0069in solid #000000;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #000000;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''IMPORTANT: Please contact IPitomy’s Technical Support Group for assistance if you feel you need to change these settings.'''&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
===== Edit CODEC Settings =====&lt;br /&gt;
&lt;br /&gt;
The following outlines the steps to set CODECs parameters.&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''Edit Extensions''' page, locate the extension that you want to update. Click the [[File:Penciledit.png]] (edit extensions) icon to the right of the name.&lt;br /&gt;
You can also '''edit multiple''' extensions by selecting (placing a checkmark) in the boxes to the left of the extensions you want to update. Click the Edit PBX Settings button located at the top right hand corner of the list.You will see the extensions that are currently being updated. Make sure that the “'''Only save the changed field'''s” box is selected.&lt;br /&gt;
&lt;br /&gt;
#The '''Extension Detail''' page appears. Select the '''Advanced''' link to open the '''Advanced Settings''' page.&lt;br /&gt;
#Make the necessary changes in the '''Allow CODECs Settings''' section of the page. Clicking Up or Down when highlighting a codec will allow you to raise or lower its usage priority.&lt;br /&gt;
#Click the [[File:Savechanges.png]] button to save the changes. The system returns you to the '''Edit Extensions''' page.&lt;br /&gt;
#Click the '''Apply Changes''' link located on the right hand corner of the page, to commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
==== Extensions Calling Permissions Section ====&lt;br /&gt;
&lt;br /&gt;
Calling permissions define the types of calls that can be sent and received from an extension and the call actions this extension can take. For example, you may want to limit who has the ability to monitor another extension. [[File:Callingpermissions.png|center|Callingpermissions.png]]&amp;lt;br/&amp;gt;The following table describes the settings for an extension’s calling permissions.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Sections/Fields'''&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Description'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow User to Control Forwarding'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If '''Enabled''' (checked), this allows the user to modify their forwarding and schedule settings when they log into the Smart Personal Console user interface.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow User to Control Follow-Me'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If '''Enabled''' (checked), this allows the user to modify their Follow-Me settings when they log into the Smart Personal Console user interface.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow User to Control Phone Key Settings'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If '''Enabled''' (checked), this allows the user to modify their Phone Key settings when they log into the Smart Personal Console user interface.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Internal Calls'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If '''Enabled''' (checked), this permits calls made from internal extensions.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow Incoming Intercom Paging'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If '''Enabled''' (checked), this permits a page to be heard through this extension.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow Outgoing Intercom Paging'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If '''Enabled''' (checked), this permits a page to be made through this extension.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow User to Forward Calls'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If '''Enabled''' (checked), this permits an extension to forward a call or voicemail message to another destination on the system.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow User to Record Calls'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If '''Enabled''' (checked), this permits the extension to record phone conversations.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow user to Listen to Others’ Calls'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If '''Enabled''' (checked), this allows the user to listen to other user’s phone conversations.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow User to Whisper'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If '''Enabled''' (checked), this allows the user to whisper to another user during a phone conversation. Whisper is similar to Listen but you can coach and only the person at the extension can hear.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow Others to Whisper'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If '''Enabled''' (checked), other extensions can Whisper to your extensions calls&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow Others to Listen'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If '''Enabled''' (checked), other extensions can Listen to your extensions calls&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow Others to Record this User’s Calls'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If '''Enabled''' (checked), other extensions can Record calls at your extension&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow Call Park'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If '''Enabled''' (checked), this permits extension to park a call.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Is Operator'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If '''Enabled''' (checked), the extension is designated as an operator. Being an operator allows the extension to control Day/Night mode overrides.&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
===== Add/Edit Calling Permissions =====&lt;br /&gt;
&lt;br /&gt;
The following outlines the steps to set calling permissions for extensions.'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''Edit Extensions''' page, locate the extension that you want to update. Click the [[File:Penciledit.png]] (edit extensions) icon to the right of the name.&lt;br /&gt;
You can also '''edit multiple''' extensions by selecting (placing a checkmark) in the boxes to the left of the extensions you want to update. Click the Edit PBX Settings button located at the top right hand corner of the list.You will see the extensions that are currently being updated. Make sure that the “'''Only save the changed field'''s” box is selected.&lt;br /&gt;
&lt;br /&gt;
#The '''Extension Detail''' page appears. Select the '''Advanced''' link to open the '''Advanced Settings''' page.&lt;br /&gt;
#Make the necessary changes in the '''Calling Permissions''' section of the page.&lt;br /&gt;
#Click the [[File:Savechanges.png]]button to save the changes. The system returns you to the '''Edit Extensions''' page.&lt;br /&gt;
#Click the '''Apply Changes''' link located on the right hand corner of the page, to commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
==== Extensions - Follow-Me Section ====&lt;br /&gt;
&lt;br /&gt;
The Follow Me feature allows the PBX to try and find a user by calling pre-configured numbers, simultaneously or in sequence of priority. Once answered, the called party is given the option to accept or reject the call. If the call is rejected, or not answered at all, the call will return to the PBX allowing the caller to leave a Voice Mail message. &lt;br /&gt;
&lt;br /&gt;
To use Follow-Me, you can point a Menu destination or a DID to route direct to Follow-Me, or you can set an extension up to use a schedule that points to the Follow-Me at a given time.  You can also use forwarding to get to Follow-Me, a typical use would be to have the NoAnswer forwarding to to Follow-Me so a call will ring an extension, and if its not answered, it can call a persons cell phone.  &lt;br /&gt;
[[File:Follow.png|alt=|none|frame|Followmesetup.png]]&lt;br /&gt;
&amp;lt;br/&amp;gt;From this window it is easy to configure numbers to be dialed by the system in order to find the user. You can adjust the Name, Number (Number to be dialed), Type of Device that is being called, Rings to wait for an answer, and you can weight the priority from 1-20 (20 being lowest 1 being highest) to define search order.&lt;br /&gt;
&lt;br /&gt;
The Use radio box allows you to easily turn on and off numbers to be dialed. Since the SPC can be accessed via remote management, this allows the users to modify and create these lists from remote locations easily. The following table details the parameters and descriptions necessary to configure the Follow-Me feature.&lt;br /&gt;
&lt;br /&gt;
The system also allows you to record your own Prompts (recordings) under '''PBX''' '''Setup'''=&amp;gt;'''Prompts''', and then use those prompts as custom messages for FollowMe feature.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{| class=&amp;quot;wikitable&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| '''Sections/Fields'''&lt;br /&gt;
| &amp;lt;center&amp;gt;'''Description'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| '''Play the Incoming Message to Caller before Starting Search'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| When Enabled (checked), the system plays the Status Prompt to the caller.&lt;br /&gt;
|-&lt;br /&gt;
| '''Record the Caller’s Name and Play it to You'''&lt;br /&gt;
| When Enabled (checked), the caller will be asked to record their name, and will announce that recorded name prior to prompting the called party to accept or reject the call.&lt;br /&gt;
|-&lt;br /&gt;
| '''Play the Unreachable Message if You Could Not Be Found'''&lt;br /&gt;
| When Enabled (checked), this will play the Sorry Prompt if the call is not answered, otherwise it goes right to the voicemail greeting.&lt;br /&gt;
|-&lt;br /&gt;
| '''Number of Seconds to Total Search Time so Caller Has Time to Listen &amp;amp; Record'''&lt;br /&gt;
| This allows you to configure how many seconds the system will spend searching for the called party. Default is 12 seconds.&lt;br /&gt;
|-&lt;br /&gt;
| '''Call From Prompt'''&lt;br /&gt;
| This plays when Record the Caller’s Name and Play it to You is enabled (checked). The system default message is “Incoming Call From” followed by the recording the caller made of their name.&lt;br /&gt;
|-&lt;br /&gt;
| '''No Recording Prompt'''&lt;br /&gt;
| This plays when Record the Caller’s Name and Play it to You is disabled (not checked). The system default message of “You have an incoming call” followed by the Options Prompt.&lt;br /&gt;
|-&lt;br /&gt;
| '''Options Prompt'''&lt;br /&gt;
| This plays after you have answered the call and prompts you to press either “1” to accept the call or “2” to reject the call. The system default message can be changed, but the options remain the same.&lt;br /&gt;
|-&lt;br /&gt;
| '''Please Hold Prompt'''&lt;br /&gt;
| This plays to the caller alerting them that the system is going to find the user they are trying to reach. The system default message of “Please hold while I try to locate the person you are calling” will play during the search process.&lt;br /&gt;
|-&lt;br /&gt;
| '''Status Prompt'''&lt;br /&gt;
| This plays the system default message of “The person you are calling is not at their desk, I will try to locate them for you”.&lt;br /&gt;
|-&lt;br /&gt;
| '''Sorry Prompt'''&lt;br /&gt;
| This plays if the person could not be reached or they reject the call. The system default message of “I’m sorry, but I was unable to locate the person you were calling”.&lt;br /&gt;
|-&lt;br /&gt;
| '''Music On Hold'''&lt;br /&gt;
| This allows you to specify a particular Music On Hold playlist to play to the caller during the search process. The system default Music on Hold (set at PBX Setup-&amp;gt;Music on Hold) will play when this parameter is set to system default.&lt;br /&gt;
|-&lt;br /&gt;
| '''Numbers'''&lt;br /&gt;
| If selected (checked), the Follow Me feature will try to either simultaneously or in sequence of priority to try and locate you by using this call list.&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
==== Add/Edit Follow Me Settings ====&lt;br /&gt;
&lt;br /&gt;
The following outlines the steps to set calling permissions for extensions.&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''Edit Extensions''' page, locate the extension that you want to update. Click the [[File:Penciledit.png]] (edit extensions) icon to the right of the name.&lt;br /&gt;
#The '''Extension Detail''' page appears. Select the '''Advanced''' link to open the '''Advanced Settings''' page.&lt;br /&gt;
#Click on the '''Numbers and Settings''' button under the '''Follow-Me''' section of the '''Advanced''' page.&lt;br /&gt;
#The '''Administrator View Follow-Me Settings/Extensions''' window appears.&lt;br /&gt;
#Make the necessary changes in the '''Follow-Me''' parameters. Click on the '''SAVE''' button to save the changes to the Follow-Me settings. A message confirming the changes will appear.&lt;br /&gt;
#Click the '''EXIT''' button to close the window.&lt;br /&gt;
#Click the [[File:Savechanges.png]] button to save the changes. The system returns you to the '''Edit Extensions''' page.&lt;br /&gt;
#Click the '''Apply Changes''' link located at the right hand corner of the top of the page, to commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
==== Multiple Simultaneous Calling ====&lt;br /&gt;
&lt;br /&gt;
See the following guide to have a call ring two locations at the same time.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:FollowMe - Multiple Simultaneous Calling Guide.pdf|File:FollowMe - Multiple Simultaneous Calling Guide.pdf]]&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=File:Crap.png&amp;diff=5168</id>
		<title>File:Crap.png</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=File:Crap.png&amp;diff=5168"/>
		<updated>2025-05-21T12:44:55Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;jw&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=Port_Forwarding&amp;diff=5167</id>
		<title>Port Forwarding</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=Port_Forwarding&amp;diff=5167"/>
		<updated>2025-05-09T16:58:05Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: added IP addresses column&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;__NOTITLE__&lt;br /&gt;
[[Category: Router Configuration]]&lt;br /&gt;
&lt;br /&gt;
== Port Forwarding&amp;lt;br/&amp;gt; ==&lt;br /&gt;
&lt;br /&gt;
The following table outlines the port forwarding information in the router that maps public IP addresses to internal IP addresses. Port forwarding must be configured to utilize features such as remote phones, SIP Providers, remote administration and branch office. IPitomy port forwarding requirements are specified below. Note you should only forward ports you intend to use.&lt;br /&gt;
&lt;br /&gt;
;Remote Phones&amp;amp;nbsp;&lt;br /&gt;
:SIP + RTP&lt;br /&gt;
;Web Access to PBX (do not forward without changing password)&amp;amp;nbsp;&lt;br /&gt;
:HTTP/HTTPS&lt;br /&gt;
;Branch Office Networking&amp;amp;nbsp;&lt;br /&gt;
:IAX2&lt;br /&gt;
;Text Messaging (via eJabbard - NOT SMS)&lt;br /&gt;
:XMPP&lt;br /&gt;
;Tech Support Access&amp;amp;nbsp;&lt;br /&gt;
:SSH - you should always forward this before you contact Technical Support&lt;br /&gt;
&lt;br /&gt;
{| class=&amp;quot;wikitable&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
! Application&lt;br /&gt;
! Port&lt;br /&gt;
! Protocol&lt;br /&gt;
! Transport Protocol&lt;br /&gt;
! LAN IP Address&lt;br /&gt;
!Assigned WAN IP&lt;br /&gt;
Addresses&lt;br /&gt;
|-&lt;br /&gt;
| Remote Management&amp;lt;br/&amp;gt;&lt;br /&gt;
| 80&lt;br /&gt;
| HTTP&lt;br /&gt;
| TCP&lt;br /&gt;
| IP Address of PBX&lt;br /&gt;
|Not Assigned&lt;br /&gt;
|-&lt;br /&gt;
| Secure Remote Management&amp;lt;br/&amp;gt;&lt;br /&gt;
| 443&amp;lt;br/&amp;gt;&lt;br /&gt;
| HTTPS&lt;br /&gt;
| TCP&lt;br /&gt;
| IP Address of PBX&amp;lt;br/&amp;gt;&lt;br /&gt;
|Not Assigned&lt;br /&gt;
|-&lt;br /&gt;
| SSH (IPitomy Support)&amp;lt;br/&amp;gt;&lt;br /&gt;
| 22&lt;br /&gt;
| SSH&amp;lt;br/&amp;gt;&lt;br /&gt;
| TCP&lt;br /&gt;
| IP Address of PBX&lt;br /&gt;
|47.206.59.221, &lt;br /&gt;
74.93.34.251&lt;br /&gt;
|-&lt;br /&gt;
| SIP Control&lt;br /&gt;
| 5060&lt;br /&gt;
| SIP&lt;br /&gt;
| UDP&lt;br /&gt;
| IP Address of PBX&lt;br /&gt;
|52.5.220.123     (SIP EAST)&lt;br /&gt;
54.200.236.200 (SIP WEST)&lt;br /&gt;
|-&lt;br /&gt;
| SIP RTP (Voice)&lt;br /&gt;
| 100000-20000&lt;br /&gt;
| RTP&lt;br /&gt;
| UDP + TCP&lt;br /&gt;
| IP Address of PBX&lt;br /&gt;
|52.5.220.123     (SIP EAST)&lt;br /&gt;
&lt;br /&gt;
54.200.236.200 (SIP WEST)&lt;br /&gt;
&lt;br /&gt;
52.202.13.217   (Media EAST)&lt;br /&gt;
&lt;br /&gt;
44.236.207.157 (Media WEST)&lt;br /&gt;
|-&lt;br /&gt;
| Chat (Text Messaging)&lt;br /&gt;
| 5222&lt;br /&gt;
| XMPP&lt;br /&gt;
| TCP&lt;br /&gt;
| IP Address of PBX&lt;br /&gt;
|Not Assigned&lt;br /&gt;
|-&lt;br /&gt;
| Branch Office&lt;br /&gt;
| 4569&lt;br /&gt;
| IAX2&lt;br /&gt;
| UDP + TCP&lt;br /&gt;
| IP Address of PBX&lt;br /&gt;
|Not Assigned&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Note that when you are configuring a port forward in your router, that many routers let you specify an external and internal port.&amp;amp;nbsp; The internal port numbers that are forwarded to the PBX must match the numbers fromt the table above.&amp;amp;nbsp; However, for SSH, HTTP, or HTTPS you may use any external port number that is available in your router.'''&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=File:CommunicatorDownload.png&amp;diff=5135</id>
		<title>File:CommunicatorDownload.png</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=File:CommunicatorDownload.png&amp;diff=5135"/>
		<updated>2025-02-26T20:16:44Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Communicator Download screenshot&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=Grandstream_410X_Setup&amp;diff=5133</id>
		<title>Grandstream 410X Setup</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=Grandstream_410X_Setup&amp;diff=5133"/>
		<updated>2025-02-07T20:35:46Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Grandstream GXW 410X Gateway and HT841 FXO FXS Gateway */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;== Grandstream GXW 410X Gateway and HT841 FXO FXS Gateway ==&lt;br /&gt;
&lt;br /&gt;
If you are installing and HT841 gateway, you will program it the same as the GXW 410X below with 2 added settings.  &lt;br /&gt;
&lt;br /&gt;
# In the SIP provider page you will change the Port to Custom and go to the fxo profile and look to see what port is manually.&lt;br /&gt;
# In the HT841 you will wet the FXO Port settings for the Hunt Group to port 1. Active, and the rest of the ports to 1.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:222222.png|alt=|frameless|600x600px]]&lt;br /&gt;
&lt;br /&gt;
[[File:HT841 ports.png|none|thumb|600x600px]][[File:Ht881.png|alt=|frameless|602x602px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Ht881-2.png|alt=|frameless|600x600px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Description&lt;br /&gt;
&lt;br /&gt;
The following configuration guide is for use setting up the Grandstream GXW410X gateways with analog circuits.&lt;br /&gt;
&lt;br /&gt;
NOTE: This guide is based upon the most current software from Grandstream. However if you have older firmware this is a link to the older guide which documents and covers the older firmware. &amp;amp;nbsp;The 4104 and 4108 use the exact same firmware as each other and have the same interface.&lt;br /&gt;
&lt;br /&gt;
[http://wiki.ipitomy.com/images/5/57/Tech_Bulletin_2011-008_Grandstream_Configuration_Guide.pdf http://wiki.ipitomy.com/images/5/57/Tech_Bulletin_2011-008_Grandstream_Configuration_Guide.pdf]&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream1.JPG|File:Gstream1.JPG]]&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Connections ===&lt;br /&gt;
&lt;br /&gt;
1. Connect each analog circuit to the desired FXO port, as above&lt;br /&gt;
&lt;br /&gt;
2. Connect the Grandstream WAN port to an available LAN port of the network switch/router being used on site.&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream2.JPG|File:Gstream2.JPG]]&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Power Up and Login ===&lt;br /&gt;
&lt;br /&gt;
1. Power up the unit and identify its assigned IP address. (Typically assigned from the DCHP server of the host router.)&lt;br /&gt;
&lt;br /&gt;
2. Use your browser to access the Grandstream by inputting the IP Address assigned to it. The IP Address assigned by your router via DHCP can be discovered several ways – the easiest of which is likely by accessing the router’s connected devices page and finding it listed there.&lt;br /&gt;
&lt;br /&gt;
3. When the Grandstream page is accessed, input the password (admin at default) and navigate to the pages below making the changes as defined.&lt;br /&gt;
&lt;br /&gt;
=== IMPORTANT: Procedure Line Setup&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
On first login to the Grandstream, you will be brought to the following page.&lt;br /&gt;
&lt;br /&gt;
[[File:GrandstreamCPT.jpg|File:GrandstreamCPT.jpg]]&lt;br /&gt;
&lt;br /&gt;
In ALL new installs once you have the device on the Network and lines connected, you should run each of these tests. The first is for Call quality and is MOST important. As this test matches line impedance which if it is in variance can cause echo and poor calls. The second allows the device to learn Call Progress Tones (CPT) and is also very important so that the Unit can learn disconnect and busy tones on the lines. NOTE: The Grandstream can learn a Disconnect Supervision Current drop and a busy tone for disconnect. Lastly is the test for CID. This is also important and additionally can detect CID presence on the line. For proper performance of the Grandstream Device these tests SHOULD NOT be skipped.&lt;br /&gt;
&lt;br /&gt;
NOTE: These tests should be performed at the site and with the lines that the unit is intended to be in service upon.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuration&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
1. Navigate to the Networks Basic Setting page and input the data as in the example.&lt;br /&gt;
&lt;br /&gt;
[[File:SIP3.jpg|File:SIP3.jpg]]&amp;lt;br/&amp;gt;2. Assign a Static IP Address. The device must be found by the IPBX regardless of incidental changes and network adjustments. For this reason its best to change the IP Address to Static and assign an address that is out of range of those assigned for DCHP subscription. (E.g. if the router will assign DCHP Addresses from 192.168.1.1 ~ 192.168.1.50 you should select an IP Address out of this range ...192.168.1.200 would work unless it is being used elsewhere.)&amp;lt;br/&amp;gt;3. Use the other information provided by the DCHP assignment process in the remaining data fields; Subnet is usually 255.255.255.0. The Default Router Address must be that of the router the same one that assigns DCHP IP Addresses. DNS should also be the router since it will direct traffic.&amp;lt;br/&amp;gt;4. Click the save button. This saves information on this page before moving on.&amp;lt;br/&amp;gt;5. Navigate to the FXO Lines page. Then Select Dialing on the left.&amp;lt;br/&amp;gt;6. Change the Stage Method(1/2)&amp;lt;br/&amp;gt;Ch1-4:1;&lt;br /&gt;
&lt;br /&gt;
[[File:Gxw410x stagemethod.PNG|File:Gxw410x stagemethod.PNG]]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;7. Navigate to Settings and then select Channels Settings on the left. Program DTMF Methods to ch1-4:2;.  Program the Unconditional Call Forward to VOIP: to include the DID (Direct Inward Dial) number(s) that are to be routed.This routing is accomplished by Profiles 1, 2 &amp;amp; 3. Usually only one is necessary.This data field is the routing of the calls that are received on this FXO circuit (It is recomended to use our example on this line which is&amp;lt;br/&amp;gt;ch1:1111;ch2:2222;ch3:3333;ch4:4444;&amp;lt;br/&amp;gt;This will create &amp;quot;virtual DIDs&amp;quot; of 1111, 2222, etc, that allow you to enter DIDs in the PBX under the SIP Provider so you can route calls to different locations based on what line they came in on.&amp;amp;nbsp; We have programmed the Grandstream to route calls that have been received to the SIP Server using Profile 1.&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream3.JPG|File:Gstream3.JPG]]&lt;br /&gt;
&lt;br /&gt;
8.Click the save button.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuration Profile ===&lt;br /&gt;
&lt;br /&gt;
[[File:SIP.jpg|File:SIP.jpg]]&lt;br /&gt;
&lt;br /&gt;
1. Navigate to the Accounts=&amp;gt;Account 1=&amp;gt;General Settings page.&amp;lt;br/&amp;gt;2. Here the SIP Server must be programmed. Set this to be the IP Address of the IPBX. In our example, the address is 192.168.2.153.&amp;lt;br/&amp;gt;3. Also be sure that the SIP Registration field is set to No.&amp;lt;br/&amp;gt;4.Click the save button.&amp;lt;br/&amp;gt;5.The Grandstream configuration is now complete. However if you wish to make changes to the dial plan allowed digits you must also program that information.&amp;lt;br/&amp;gt;Note: At default the Grandstream allows only digits 0-9 to be sent to the connected PSTN circuit. If you want to use PSTN features like call forward you will need to change the Outgoing Dial Plan field to;&amp;lt;br/&amp;gt;{x+ | [x*]+}&lt;br /&gt;
&lt;br /&gt;
[[File:Sip2.jpg|File:Sip2.jpg]]&amp;lt;br/&amp;gt;to be able to send a * to outside IVR's&amp;lt;br/&amp;gt;6. When programming is complete in the Grandstream, click submit will commit the changes saved thus far to memory and make them operational. Continue to Procedure Configuring the IPitomy IP PBX&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream5.JPG|File:Gstream5.JPG]]&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuring the IPitomy IP PBX for Grandstream GXW 410x ===&lt;br /&gt;
&lt;br /&gt;
1.In the IPitomy IPBX set the fields as you see them below... using the Static IP Address assigned to the Procedure—Configuration step previously completed.. (In our example we assigned the Grandstream an IP Address of 192.168.2.9. This becomes the ―Host)&lt;br /&gt;
&lt;br /&gt;
2. Click “Save Changes”&lt;br /&gt;
&lt;br /&gt;
3. Then click on theApply Changes (upper right) to make these settings operational in the Ipitomy IPBX.&lt;br /&gt;
&lt;br /&gt;
4.Test the operation. Make a call into each of the Grandstream ports that have circuits and assure that they are being routed as defined in Call Routing—Incoming.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;5.Test the operation. Make a call at an Ipitomy extension using a calling pattern as defined in Call Routing Outgoing to assure that the call that should be placed over the Grandstream ports are placed.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuring Optional Outbound Routing Methods ===&lt;br /&gt;
&lt;br /&gt;
The Grandstream allows for two methods of outbound dialing at the same time.&lt;br /&gt;
&lt;br /&gt;
Round-Robin (Linear Hunt)&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;If you add the trunk to an outbound route in the PBX, the Grandstream will follow the Round-Robin rules, which are found on the Channels page. By default it will start at Line&lt;br /&gt;
&lt;br /&gt;
1, and move on until it finds an available channel to dial outbound.If you add the trunk to an outbound route and configure Prefix Digits to 99X where X is the port on the Grandstream you wish to use when placing this call (e.g. 991 is line 1). By doing this you can configure the calls to route out a particular line, or a different order if you add the trunk multiple times and prefix accordingly (992, 994, 991, 993, etc). The code of 99 can be changed on the Channels page in the Grandstream.&lt;br /&gt;
&lt;br /&gt;
1. Navigate to the Channels page&lt;br /&gt;
&lt;br /&gt;
2. Go to the Port Scheduling Schema (Voip-&amp;gt;PSTN) section and input the code or codes that you wish (99x)&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configure Multiple Line Groups on Single Grandstream ===&lt;br /&gt;
&lt;br /&gt;
'''1 -''' In order to put the lines into separate outbound groups you will need to configure each round robin group under FXO Lines =&amp;gt; Dialing.  In this example lines 1-3 are in the first group and line 4 is standalone.&lt;br /&gt;
&lt;br /&gt;
[[File:GS RR Separate.PNG|File:GS RR Separate.PNG]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''2 -''' After setting up the round robin groups you now have to configure the local SIP port for the groups.  In this example channels 1-3 are using 5060 and incrementing by 2 (this is default behavior for multiple channels in a group, any call sent to 5060 will use the first available line in the group), channel 4 is alone listening on port 5160.&lt;br /&gt;
&lt;br /&gt;
[[File:GS RR Ports.PNG]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''3 -''' After configuring the local SIP port you now need to configure the account SIP User ID, this will allow the PBX to differentiate between each line group on incoming calls.  Lines 1-3 will give a user ID of 0001, and line 4 will give a user ID of 0002.&lt;br /&gt;
&lt;br /&gt;
[[File:GS RR Account.PNG]]&lt;br /&gt;
&lt;br /&gt;
'''4 -''' On the PBX side we will need to create two trunks, both pointing to the same IP address of the Grandstream, but each will have a different user.  '''Be absolutely certain you configure the trunk with Insecure = No''', this setting will tell the PBX to consider the user when taking the call.  Notice that on the 2nd trunk we've added the correct port number for line 4 and the correct User associated with line 4.  These two settings are what make the 2nd trunk use line 4, the first trunk is using the default port of 5060 and therefore only needs the user to distinguish incoming calls.&lt;br /&gt;
&lt;br /&gt;
[[File:PBX RR Trunk.PNG]]&lt;br /&gt;
&lt;br /&gt;
Outgoing calls will now be sent to port 5060 on the Grandstream to use lines 1-3, calls sent to 5160 will use line 4.  Incoming calls on lines 1-3 will hit the PBX with the user of 0001 which will select the first trunk, incoming calls on line 4 will hit the PBX with user 0002 selecting the 2nd trunk.&lt;br /&gt;
&lt;br /&gt;
== Remote Grandstream Configuration ==&lt;br /&gt;
&lt;br /&gt;
In order to get a Grandstream 410x working as a remote device, you must make a few changes from the standard, local configuration. Use this guide in conjunction with the standard guide.&lt;br /&gt;
&lt;br /&gt;
=== Grandstream Changes ===&lt;br /&gt;
&lt;br /&gt;
#On the Accounts=&amp;gt;Account X=&amp;gt;SIP Settings page, set SIP Registration to YES&lt;br /&gt;
#Under the User Accounts page, configure one channel with login credentials:&lt;br /&gt;
#Set the SIP server in the Profile to the PBX Public IP.&lt;br /&gt;
&lt;br /&gt;
[[File:Gxw410x remotesettings.PNG|File:Gxw410x remotesettings.PNG]]&lt;br /&gt;
&lt;br /&gt;
=== PBX SIP Provider Changes&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
#Instead of entering an IP address for Host, set to “dynamic”&lt;br /&gt;
#Set Register and Authentication to YES&lt;br /&gt;
#Set Username and Name to the value entered as SIP User ID and Authenticate ID in the grandstream&lt;br /&gt;
#Set Secret to the value entered for Authen Password in the grandstream&lt;br /&gt;
&lt;br /&gt;
Note: If the grandstream is not the only device at the remote site, then all remote SIP devices will need to have Can Reinvite set to YES.&lt;br /&gt;
&lt;br /&gt;
== Troubleshooting ==&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Outbound Dialing ===&lt;br /&gt;
&lt;br /&gt;
If you having trouble dialing outbound make the following changes on the FXO Lines page:&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;1. Tweak the Disconnect Threshold from 100 to 300ms.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;2. Tweak the Minimum Delay Before Dialing Out from 500 to 750ms.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Call Quality ===&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;If you are having call quality issues try the following changes on the Channels page:&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;1.Set Silence Suppression from YES to NO.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;2. Set Echo Cancellation from YES to NO&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Call Volume ===&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;If you are having issue with call volume try the following changes on the Channels page:&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;1.Increase/Decrease Tx to PSTN Audio Gain by increments of 3 for issues with external party volume.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;2. Increase/Decrease Rx from PSTN Audio Gain by increments of 3 for issues with internal party volume&lt;br /&gt;
&lt;br /&gt;
&amp;lt;span style=&amp;quot;font-size:large&amp;quot;&amp;gt;'''&amp;amp;nbsp;Procedure—Troubleshooting—Call Buzzing Noise'''&amp;lt;/span&amp;gt;&lt;br /&gt;
&lt;br /&gt;
If you are having issues with a buzz heard prior to a Menu prompt; try upgrading the Grandstream firmware:&lt;br /&gt;
&lt;br /&gt;
1. Navigate to Advanced Settings page&lt;br /&gt;
&lt;br /&gt;
2. Ensure HTTP is selected for the method to upgrade&lt;br /&gt;
&lt;br /&gt;
3. Set Firmware Server Path: to firmware.grandstream.com&lt;br /&gt;
&lt;br /&gt;
4. Set Automatic Upgrade to YES&lt;br /&gt;
&lt;br /&gt;
5. Set Allow DHCP Option 66 to override server to No Click Update at the bottom of the page&lt;br /&gt;
&lt;br /&gt;
6. Click Reboot The upgrade may take as long as 20min when done through the internet, so allow plenty of time for this. While upgrading the LED will blink. When the LED returns to normal, the device has completed its upgrade&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Restore Factory Default ===&lt;br /&gt;
&lt;br /&gt;
To Restore Factory Defaults:&lt;br /&gt;
&lt;br /&gt;
1. While powered up, hold the recessed Reset button in for 7+ seconds&lt;br /&gt;
&lt;br /&gt;
'''&amp;lt;span style=&amp;quot;font-size:large&amp;quot;&amp;gt;[[File:Gstream6.JPG|File:Gstream6.JPG]]&amp;amp;nbsp;&amp;lt;/span&amp;gt;'''&lt;br /&gt;
&lt;br /&gt;
'''&amp;lt;span style=&amp;quot;font-size:large&amp;quot;&amp;gt;Procedure Setup of QOS&amp;lt;/span&amp;gt;'''&lt;br /&gt;
&lt;br /&gt;
1. Access the IP of the Grandstream.&lt;br /&gt;
&lt;br /&gt;
2. Navigate in the Grandstream Web Admin to Networks – Advanced Settings&lt;br /&gt;
&lt;br /&gt;
3. Set the Layer 3 Diff Serv Value to 24 Which is CS3 or in Binary 011000&lt;br /&gt;
&lt;br /&gt;
4. Submit your changes and reboot&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;{{:Grandstream_FXO_FAQ}}&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=File:222222.png&amp;diff=5132</id>
		<title>File:222222.png</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=File:222222.png&amp;diff=5132"/>
		<updated>2025-02-07T20:35:27Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;jw&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=Grandstream_410X_Setup&amp;diff=5131</id>
		<title>Grandstream 410X Setup</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=Grandstream_410X_Setup&amp;diff=5131"/>
		<updated>2025-02-07T15:48:05Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Grandstream GXW 410X Gateway and HT841 FXO FXS Gateway */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;== Grandstream GXW 410X Gateway and HT841 FXO FXS Gateway ==&lt;br /&gt;
&lt;br /&gt;
If you are installing and HT841 gateway, you will program it the same as the GXW 410X below with 2 added settings.  &lt;br /&gt;
&lt;br /&gt;
# In the SIP provider page you will change the Port to Custom and make it 6062.&lt;br /&gt;
# In the HT841 you will wet the FXO Port settings for the Hunt Group to port 1. Active, and the rest of the ports to 1.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:HT841.png|frameless|600x600px]]&lt;br /&gt;
&lt;br /&gt;
[[File:HT841 ports.png|none|thumb|600x600px]][[File:Ht881.png|alt=|frameless|602x602px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Ht881-2.png|alt=|frameless|600x600px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Description&lt;br /&gt;
&lt;br /&gt;
The following configuration guide is for use setting up the Grandstream GXW410X gateways with analog circuits.&lt;br /&gt;
&lt;br /&gt;
NOTE: This guide is based upon the most current software from Grandstream. However if you have older firmware this is a link to the older guide which documents and covers the older firmware. &amp;amp;nbsp;The 4104 and 4108 use the exact same firmware as each other and have the same interface.&lt;br /&gt;
&lt;br /&gt;
[http://wiki.ipitomy.com/images/5/57/Tech_Bulletin_2011-008_Grandstream_Configuration_Guide.pdf http://wiki.ipitomy.com/images/5/57/Tech_Bulletin_2011-008_Grandstream_Configuration_Guide.pdf]&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream1.JPG|File:Gstream1.JPG]]&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Connections ===&lt;br /&gt;
&lt;br /&gt;
1. Connect each analog circuit to the desired FXO port, as above&lt;br /&gt;
&lt;br /&gt;
2. Connect the Grandstream WAN port to an available LAN port of the network switch/router being used on site.&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream2.JPG|File:Gstream2.JPG]]&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Power Up and Login ===&lt;br /&gt;
&lt;br /&gt;
1. Power up the unit and identify its assigned IP address. (Typically assigned from the DCHP server of the host router.)&lt;br /&gt;
&lt;br /&gt;
2. Use your browser to access the Grandstream by inputting the IP Address assigned to it. The IP Address assigned by your router via DHCP can be discovered several ways – the easiest of which is likely by accessing the router’s connected devices page and finding it listed there.&lt;br /&gt;
&lt;br /&gt;
3. When the Grandstream page is accessed, input the password (admin at default) and navigate to the pages below making the changes as defined.&lt;br /&gt;
&lt;br /&gt;
=== IMPORTANT: Procedure Line Setup&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
On first login to the Grandstream, you will be brought to the following page.&lt;br /&gt;
&lt;br /&gt;
[[File:GrandstreamCPT.jpg|File:GrandstreamCPT.jpg]]&lt;br /&gt;
&lt;br /&gt;
In ALL new installs once you have the device on the Network and lines connected, you should run each of these tests. The first is for Call quality and is MOST important. As this test matches line impedance which if it is in variance can cause echo and poor calls. The second allows the device to learn Call Progress Tones (CPT) and is also very important so that the Unit can learn disconnect and busy tones on the lines. NOTE: The Grandstream can learn a Disconnect Supervision Current drop and a busy tone for disconnect. Lastly is the test for CID. This is also important and additionally can detect CID presence on the line. For proper performance of the Grandstream Device these tests SHOULD NOT be skipped.&lt;br /&gt;
&lt;br /&gt;
NOTE: These tests should be performed at the site and with the lines that the unit is intended to be in service upon.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuration&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
1. Navigate to the Networks Basic Setting page and input the data as in the example.&lt;br /&gt;
&lt;br /&gt;
[[File:SIP3.jpg|File:SIP3.jpg]]&amp;lt;br/&amp;gt;2. Assign a Static IP Address. The device must be found by the IPBX regardless of incidental changes and network adjustments. For this reason its best to change the IP Address to Static and assign an address that is out of range of those assigned for DCHP subscription. (E.g. if the router will assign DCHP Addresses from 192.168.1.1 ~ 192.168.1.50 you should select an IP Address out of this range ...192.168.1.200 would work unless it is being used elsewhere.)&amp;lt;br/&amp;gt;3. Use the other information provided by the DCHP assignment process in the remaining data fields; Subnet is usually 255.255.255.0. The Default Router Address must be that of the router the same one that assigns DCHP IP Addresses. DNS should also be the router since it will direct traffic.&amp;lt;br/&amp;gt;4. Click the save button. This saves information on this page before moving on.&amp;lt;br/&amp;gt;5. Navigate to the FXO Lines page. Then Select Dialing on the left.&amp;lt;br/&amp;gt;6. Change the Stage Method(1/2)&amp;lt;br/&amp;gt;Ch1-4:1;&lt;br /&gt;
&lt;br /&gt;
[[File:Gxw410x stagemethod.PNG|File:Gxw410x stagemethod.PNG]]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;7. Navigate to Settings and then select Channels Settings on the left. Program DTMF Methods to ch1-4:2;.  Program the Unconditional Call Forward to VOIP: to include the DID (Direct Inward Dial) number(s) that are to be routed.This routing is accomplished by Profiles 1, 2 &amp;amp; 3. Usually only one is necessary.This data field is the routing of the calls that are received on this FXO circuit (It is recomended to use our example on this line which is&amp;lt;br/&amp;gt;ch1:1111;ch2:2222;ch3:3333;ch4:4444;&amp;lt;br/&amp;gt;This will create &amp;quot;virtual DIDs&amp;quot; of 1111, 2222, etc, that allow you to enter DIDs in the PBX under the SIP Provider so you can route calls to different locations based on what line they came in on.&amp;amp;nbsp; We have programmed the Grandstream to route calls that have been received to the SIP Server using Profile 1.&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream3.JPG|File:Gstream3.JPG]]&lt;br /&gt;
&lt;br /&gt;
8.Click the save button.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuration Profile ===&lt;br /&gt;
&lt;br /&gt;
[[File:SIP.jpg|File:SIP.jpg]]&lt;br /&gt;
&lt;br /&gt;
1. Navigate to the Accounts=&amp;gt;Account 1=&amp;gt;General Settings page.&amp;lt;br/&amp;gt;2. Here the SIP Server must be programmed. Set this to be the IP Address of the IPBX. In our example, the address is 192.168.2.153.&amp;lt;br/&amp;gt;3. Also be sure that the SIP Registration field is set to No.&amp;lt;br/&amp;gt;4.Click the save button.&amp;lt;br/&amp;gt;5.The Grandstream configuration is now complete. However if you wish to make changes to the dial plan allowed digits you must also program that information.&amp;lt;br/&amp;gt;Note: At default the Grandstream allows only digits 0-9 to be sent to the connected PSTN circuit. If you want to use PSTN features like call forward you will need to change the Outgoing Dial Plan field to;&amp;lt;br/&amp;gt;{x+ | [x*]+}&lt;br /&gt;
&lt;br /&gt;
[[File:Sip2.jpg|File:Sip2.jpg]]&amp;lt;br/&amp;gt;to be able to send a * to outside IVR's&amp;lt;br/&amp;gt;6. When programming is complete in the Grandstream, click submit will commit the changes saved thus far to memory and make them operational. Continue to Procedure Configuring the IPitomy IP PBX&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream5.JPG|File:Gstream5.JPG]]&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuring the IPitomy IP PBX for Grandstream GXW 410x ===&lt;br /&gt;
&lt;br /&gt;
1.In the IPitomy IPBX set the fields as you see them below... using the Static IP Address assigned to the Procedure—Configuration step previously completed.. (In our example we assigned the Grandstream an IP Address of 192.168.2.9. This becomes the ―Host)&lt;br /&gt;
&lt;br /&gt;
2. Click “Save Changes”&lt;br /&gt;
&lt;br /&gt;
3. Then click on theApply Changes (upper right) to make these settings operational in the Ipitomy IPBX.&lt;br /&gt;
&lt;br /&gt;
4.Test the operation. Make a call into each of the Grandstream ports that have circuits and assure that they are being routed as defined in Call Routing—Incoming.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;5.Test the operation. Make a call at an Ipitomy extension using a calling pattern as defined in Call Routing Outgoing to assure that the call that should be placed over the Grandstream ports are placed.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuring Optional Outbound Routing Methods ===&lt;br /&gt;
&lt;br /&gt;
The Grandstream allows for two methods of outbound dialing at the same time.&lt;br /&gt;
&lt;br /&gt;
Round-Robin (Linear Hunt)&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;If you add the trunk to an outbound route in the PBX, the Grandstream will follow the Round-Robin rules, which are found on the Channels page. By default it will start at Line&lt;br /&gt;
&lt;br /&gt;
1, and move on until it finds an available channel to dial outbound.If you add the trunk to an outbound route and configure Prefix Digits to 99X where X is the port on the Grandstream you wish to use when placing this call (e.g. 991 is line 1). By doing this you can configure the calls to route out a particular line, or a different order if you add the trunk multiple times and prefix accordingly (992, 994, 991, 993, etc). The code of 99 can be changed on the Channels page in the Grandstream.&lt;br /&gt;
&lt;br /&gt;
1. Navigate to the Channels page&lt;br /&gt;
&lt;br /&gt;
2. Go to the Port Scheduling Schema (Voip-&amp;gt;PSTN) section and input the code or codes that you wish (99x)&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configure Multiple Line Groups on Single Grandstream ===&lt;br /&gt;
&lt;br /&gt;
'''1 -''' In order to put the lines into separate outbound groups you will need to configure each round robin group under FXO Lines =&amp;gt; Dialing.  In this example lines 1-3 are in the first group and line 4 is standalone.&lt;br /&gt;
&lt;br /&gt;
[[File:GS RR Separate.PNG|File:GS RR Separate.PNG]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''2 -''' After setting up the round robin groups you now have to configure the local SIP port for the groups.  In this example channels 1-3 are using 5060 and incrementing by 2 (this is default behavior for multiple channels in a group, any call sent to 5060 will use the first available line in the group), channel 4 is alone listening on port 5160.&lt;br /&gt;
&lt;br /&gt;
[[File:GS RR Ports.PNG]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''3 -''' After configuring the local SIP port you now need to configure the account SIP User ID, this will allow the PBX to differentiate between each line group on incoming calls.  Lines 1-3 will give a user ID of 0001, and line 4 will give a user ID of 0002.&lt;br /&gt;
&lt;br /&gt;
[[File:GS RR Account.PNG]]&lt;br /&gt;
&lt;br /&gt;
'''4 -''' On the PBX side we will need to create two trunks, both pointing to the same IP address of the Grandstream, but each will have a different user.  '''Be absolutely certain you configure the trunk with Insecure = No''', this setting will tell the PBX to consider the user when taking the call.  Notice that on the 2nd trunk we've added the correct port number for line 4 and the correct User associated with line 4.  These two settings are what make the 2nd trunk use line 4, the first trunk is using the default port of 5060 and therefore only needs the user to distinguish incoming calls.&lt;br /&gt;
&lt;br /&gt;
[[File:PBX RR Trunk.PNG]]&lt;br /&gt;
&lt;br /&gt;
Outgoing calls will now be sent to port 5060 on the Grandstream to use lines 1-3, calls sent to 5160 will use line 4.  Incoming calls on lines 1-3 will hit the PBX with the user of 0001 which will select the first trunk, incoming calls on line 4 will hit the PBX with user 0002 selecting the 2nd trunk.&lt;br /&gt;
&lt;br /&gt;
== Remote Grandstream Configuration ==&lt;br /&gt;
&lt;br /&gt;
In order to get a Grandstream 410x working as a remote device, you must make a few changes from the standard, local configuration. Use this guide in conjunction with the standard guide.&lt;br /&gt;
&lt;br /&gt;
=== Grandstream Changes ===&lt;br /&gt;
&lt;br /&gt;
#On the Accounts=&amp;gt;Account X=&amp;gt;SIP Settings page, set SIP Registration to YES&lt;br /&gt;
#Under the User Accounts page, configure one channel with login credentials:&lt;br /&gt;
#Set the SIP server in the Profile to the PBX Public IP.&lt;br /&gt;
&lt;br /&gt;
[[File:Gxw410x remotesettings.PNG|File:Gxw410x remotesettings.PNG]]&lt;br /&gt;
&lt;br /&gt;
=== PBX SIP Provider Changes&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
#Instead of entering an IP address for Host, set to “dynamic”&lt;br /&gt;
#Set Register and Authentication to YES&lt;br /&gt;
#Set Username and Name to the value entered as SIP User ID and Authenticate ID in the grandstream&lt;br /&gt;
#Set Secret to the value entered for Authen Password in the grandstream&lt;br /&gt;
&lt;br /&gt;
Note: If the grandstream is not the only device at the remote site, then all remote SIP devices will need to have Can Reinvite set to YES.&lt;br /&gt;
&lt;br /&gt;
== Troubleshooting ==&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Outbound Dialing ===&lt;br /&gt;
&lt;br /&gt;
If you having trouble dialing outbound make the following changes on the FXO Lines page:&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;1. Tweak the Disconnect Threshold from 100 to 300ms.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;2. Tweak the Minimum Delay Before Dialing Out from 500 to 750ms.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Call Quality ===&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;If you are having call quality issues try the following changes on the Channels page:&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;1.Set Silence Suppression from YES to NO.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;2. Set Echo Cancellation from YES to NO&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Call Volume ===&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;If you are having issue with call volume try the following changes on the Channels page:&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;1.Increase/Decrease Tx to PSTN Audio Gain by increments of 3 for issues with external party volume.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;2. Increase/Decrease Rx from PSTN Audio Gain by increments of 3 for issues with internal party volume&lt;br /&gt;
&lt;br /&gt;
&amp;lt;span style=&amp;quot;font-size:large&amp;quot;&amp;gt;'''&amp;amp;nbsp;Procedure—Troubleshooting—Call Buzzing Noise'''&amp;lt;/span&amp;gt;&lt;br /&gt;
&lt;br /&gt;
If you are having issues with a buzz heard prior to a Menu prompt; try upgrading the Grandstream firmware:&lt;br /&gt;
&lt;br /&gt;
1. Navigate to Advanced Settings page&lt;br /&gt;
&lt;br /&gt;
2. Ensure HTTP is selected for the method to upgrade&lt;br /&gt;
&lt;br /&gt;
3. Set Firmware Server Path: to firmware.grandstream.com&lt;br /&gt;
&lt;br /&gt;
4. Set Automatic Upgrade to YES&lt;br /&gt;
&lt;br /&gt;
5. Set Allow DHCP Option 66 to override server to No Click Update at the bottom of the page&lt;br /&gt;
&lt;br /&gt;
6. Click Reboot The upgrade may take as long as 20min when done through the internet, so allow plenty of time for this. While upgrading the LED will blink. When the LED returns to normal, the device has completed its upgrade&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Restore Factory Default ===&lt;br /&gt;
&lt;br /&gt;
To Restore Factory Defaults:&lt;br /&gt;
&lt;br /&gt;
1. While powered up, hold the recessed Reset button in for 7+ seconds&lt;br /&gt;
&lt;br /&gt;
'''&amp;lt;span style=&amp;quot;font-size:large&amp;quot;&amp;gt;[[File:Gstream6.JPG|File:Gstream6.JPG]]&amp;amp;nbsp;&amp;lt;/span&amp;gt;'''&lt;br /&gt;
&lt;br /&gt;
'''&amp;lt;span style=&amp;quot;font-size:large&amp;quot;&amp;gt;Procedure Setup of QOS&amp;lt;/span&amp;gt;'''&lt;br /&gt;
&lt;br /&gt;
1. Access the IP of the Grandstream.&lt;br /&gt;
&lt;br /&gt;
2. Navigate in the Grandstream Web Admin to Networks – Advanced Settings&lt;br /&gt;
&lt;br /&gt;
3. Set the Layer 3 Diff Serv Value to 24 Which is CS3 or in Binary 011000&lt;br /&gt;
&lt;br /&gt;
4. Submit your changes and reboot&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;{{:Grandstream_FXO_FAQ}}&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=Grandstream_410X_Setup&amp;diff=5130</id>
		<title>Grandstream 410X Setup</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=Grandstream_410X_Setup&amp;diff=5130"/>
		<updated>2025-02-07T15:47:22Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Grandstream GXW 410X Gateway and HT841 FXO FXS Gateway */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;== Grandstream GXW 410X Gateway and HT841 FXO FXS Gateway ==&lt;br /&gt;
&lt;br /&gt;
If you are installing and HT841 gateway, you will program it the same as the GXW 410X below with 2 added settings.  &lt;br /&gt;
&lt;br /&gt;
# In the SIP provider page you will change the Port to Custom and make it 6062.&lt;br /&gt;
# In the HT841 you will wet the FXO Port settings for the Hunt Group to port 1. Active, and the rest of the ports to 1.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:HT841.png|frameless|600x600px]]&lt;br /&gt;
&lt;br /&gt;
[[File:HT841 ports.png|none|thumb|600x600px]]&lt;br /&gt;
[[File:Ht881.png|left|frameless|602x602px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Ht881-2.png|alt=|frameless|600x600px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Description&lt;br /&gt;
&lt;br /&gt;
The following configuration guide is for use setting up the Grandstream GXW410X gateways with analog circuits.&lt;br /&gt;
&lt;br /&gt;
NOTE: This guide is based upon the most current software from Grandstream. However if you have older firmware this is a link to the older guide which documents and covers the older firmware. &amp;amp;nbsp;The 4104 and 4108 use the exact same firmware as each other and have the same interface.&lt;br /&gt;
&lt;br /&gt;
[http://wiki.ipitomy.com/images/5/57/Tech_Bulletin_2011-008_Grandstream_Configuration_Guide.pdf http://wiki.ipitomy.com/images/5/57/Tech_Bulletin_2011-008_Grandstream_Configuration_Guide.pdf]&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream1.JPG|File:Gstream1.JPG]]&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Connections ===&lt;br /&gt;
&lt;br /&gt;
1. Connect each analog circuit to the desired FXO port, as above&lt;br /&gt;
&lt;br /&gt;
2. Connect the Grandstream WAN port to an available LAN port of the network switch/router being used on site.&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream2.JPG|File:Gstream2.JPG]]&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Power Up and Login ===&lt;br /&gt;
&lt;br /&gt;
1. Power up the unit and identify its assigned IP address. (Typically assigned from the DCHP server of the host router.)&lt;br /&gt;
&lt;br /&gt;
2. Use your browser to access the Grandstream by inputting the IP Address assigned to it. The IP Address assigned by your router via DHCP can be discovered several ways – the easiest of which is likely by accessing the router’s connected devices page and finding it listed there.&lt;br /&gt;
&lt;br /&gt;
3. When the Grandstream page is accessed, input the password (admin at default) and navigate to the pages below making the changes as defined.&lt;br /&gt;
&lt;br /&gt;
=== IMPORTANT: Procedure Line Setup&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
On first login to the Grandstream, you will be brought to the following page.&lt;br /&gt;
&lt;br /&gt;
[[File:GrandstreamCPT.jpg|File:GrandstreamCPT.jpg]]&lt;br /&gt;
&lt;br /&gt;
In ALL new installs once you have the device on the Network and lines connected, you should run each of these tests. The first is for Call quality and is MOST important. As this test matches line impedance which if it is in variance can cause echo and poor calls. The second allows the device to learn Call Progress Tones (CPT) and is also very important so that the Unit can learn disconnect and busy tones on the lines. NOTE: The Grandstream can learn a Disconnect Supervision Current drop and a busy tone for disconnect. Lastly is the test for CID. This is also important and additionally can detect CID presence on the line. For proper performance of the Grandstream Device these tests SHOULD NOT be skipped.&lt;br /&gt;
&lt;br /&gt;
NOTE: These tests should be performed at the site and with the lines that the unit is intended to be in service upon.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuration&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
1. Navigate to the Networks Basic Setting page and input the data as in the example.&lt;br /&gt;
&lt;br /&gt;
[[File:SIP3.jpg|File:SIP3.jpg]]&amp;lt;br/&amp;gt;2. Assign a Static IP Address. The device must be found by the IPBX regardless of incidental changes and network adjustments. For this reason its best to change the IP Address to Static and assign an address that is out of range of those assigned for DCHP subscription. (E.g. if the router will assign DCHP Addresses from 192.168.1.1 ~ 192.168.1.50 you should select an IP Address out of this range ...192.168.1.200 would work unless it is being used elsewhere.)&amp;lt;br/&amp;gt;3. Use the other information provided by the DCHP assignment process in the remaining data fields; Subnet is usually 255.255.255.0. The Default Router Address must be that of the router the same one that assigns DCHP IP Addresses. DNS should also be the router since it will direct traffic.&amp;lt;br/&amp;gt;4. Click the save button. This saves information on this page before moving on.&amp;lt;br/&amp;gt;5. Navigate to the FXO Lines page. Then Select Dialing on the left.&amp;lt;br/&amp;gt;6. Change the Stage Method(1/2)&amp;lt;br/&amp;gt;Ch1-4:1;&lt;br /&gt;
&lt;br /&gt;
[[File:Gxw410x stagemethod.PNG|File:Gxw410x stagemethod.PNG]]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;7. Navigate to Settings and then select Channels Settings on the left. Program DTMF Methods to ch1-4:2;.  Program the Unconditional Call Forward to VOIP: to include the DID (Direct Inward Dial) number(s) that are to be routed.This routing is accomplished by Profiles 1, 2 &amp;amp; 3. Usually only one is necessary.This data field is the routing of the calls that are received on this FXO circuit (It is recomended to use our example on this line which is&amp;lt;br/&amp;gt;ch1:1111;ch2:2222;ch3:3333;ch4:4444;&amp;lt;br/&amp;gt;This will create &amp;quot;virtual DIDs&amp;quot; of 1111, 2222, etc, that allow you to enter DIDs in the PBX under the SIP Provider so you can route calls to different locations based on what line they came in on.&amp;amp;nbsp; We have programmed the Grandstream to route calls that have been received to the SIP Server using Profile 1.&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream3.JPG|File:Gstream3.JPG]]&lt;br /&gt;
&lt;br /&gt;
8.Click the save button.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuration Profile ===&lt;br /&gt;
&lt;br /&gt;
[[File:SIP.jpg|File:SIP.jpg]]&lt;br /&gt;
&lt;br /&gt;
1. Navigate to the Accounts=&amp;gt;Account 1=&amp;gt;General Settings page.&amp;lt;br/&amp;gt;2. Here the SIP Server must be programmed. Set this to be the IP Address of the IPBX. In our example, the address is 192.168.2.153.&amp;lt;br/&amp;gt;3. Also be sure that the SIP Registration field is set to No.&amp;lt;br/&amp;gt;4.Click the save button.&amp;lt;br/&amp;gt;5.The Grandstream configuration is now complete. However if you wish to make changes to the dial plan allowed digits you must also program that information.&amp;lt;br/&amp;gt;Note: At default the Grandstream allows only digits 0-9 to be sent to the connected PSTN circuit. If you want to use PSTN features like call forward you will need to change the Outgoing Dial Plan field to;&amp;lt;br/&amp;gt;{x+ | [x*]+}&lt;br /&gt;
&lt;br /&gt;
[[File:Sip2.jpg|File:Sip2.jpg]]&amp;lt;br/&amp;gt;to be able to send a * to outside IVR's&amp;lt;br/&amp;gt;6. When programming is complete in the Grandstream, click submit will commit the changes saved thus far to memory and make them operational. Continue to Procedure Configuring the IPitomy IP PBX&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream5.JPG|File:Gstream5.JPG]]&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuring the IPitomy IP PBX for Grandstream GXW 410x ===&lt;br /&gt;
&lt;br /&gt;
1.In the IPitomy IPBX set the fields as you see them below... using the Static IP Address assigned to the Procedure—Configuration step previously completed.. (In our example we assigned the Grandstream an IP Address of 192.168.2.9. This becomes the ―Host)&lt;br /&gt;
&lt;br /&gt;
2. Click “Save Changes”&lt;br /&gt;
&lt;br /&gt;
3. Then click on theApply Changes (upper right) to make these settings operational in the Ipitomy IPBX.&lt;br /&gt;
&lt;br /&gt;
4.Test the operation. Make a call into each of the Grandstream ports that have circuits and assure that they are being routed as defined in Call Routing—Incoming.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;5.Test the operation. Make a call at an Ipitomy extension using a calling pattern as defined in Call Routing Outgoing to assure that the call that should be placed over the Grandstream ports are placed.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuring Optional Outbound Routing Methods ===&lt;br /&gt;
&lt;br /&gt;
The Grandstream allows for two methods of outbound dialing at the same time.&lt;br /&gt;
&lt;br /&gt;
Round-Robin (Linear Hunt)&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;If you add the trunk to an outbound route in the PBX, the Grandstream will follow the Round-Robin rules, which are found on the Channels page. By default it will start at Line&lt;br /&gt;
&lt;br /&gt;
1, and move on until it finds an available channel to dial outbound.If you add the trunk to an outbound route and configure Prefix Digits to 99X where X is the port on the Grandstream you wish to use when placing this call (e.g. 991 is line 1). By doing this you can configure the calls to route out a particular line, or a different order if you add the trunk multiple times and prefix accordingly (992, 994, 991, 993, etc). The code of 99 can be changed on the Channels page in the Grandstream.&lt;br /&gt;
&lt;br /&gt;
1. Navigate to the Channels page&lt;br /&gt;
&lt;br /&gt;
2. Go to the Port Scheduling Schema (Voip-&amp;gt;PSTN) section and input the code or codes that you wish (99x)&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configure Multiple Line Groups on Single Grandstream ===&lt;br /&gt;
&lt;br /&gt;
'''1 -''' In order to put the lines into separate outbound groups you will need to configure each round robin group under FXO Lines =&amp;gt; Dialing.  In this example lines 1-3 are in the first group and line 4 is standalone.&lt;br /&gt;
&lt;br /&gt;
[[File:GS RR Separate.PNG|File:GS RR Separate.PNG]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''2 -''' After setting up the round robin groups you now have to configure the local SIP port for the groups.  In this example channels 1-3 are using 5060 and incrementing by 2 (this is default behavior for multiple channels in a group, any call sent to 5060 will use the first available line in the group), channel 4 is alone listening on port 5160.&lt;br /&gt;
&lt;br /&gt;
[[File:GS RR Ports.PNG]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''3 -''' After configuring the local SIP port you now need to configure the account SIP User ID, this will allow the PBX to differentiate between each line group on incoming calls.  Lines 1-3 will give a user ID of 0001, and line 4 will give a user ID of 0002.&lt;br /&gt;
&lt;br /&gt;
[[File:GS RR Account.PNG]]&lt;br /&gt;
&lt;br /&gt;
'''4 -''' On the PBX side we will need to create two trunks, both pointing to the same IP address of the Grandstream, but each will have a different user.  '''Be absolutely certain you configure the trunk with Insecure = No''', this setting will tell the PBX to consider the user when taking the call.  Notice that on the 2nd trunk we've added the correct port number for line 4 and the correct User associated with line 4.  These two settings are what make the 2nd trunk use line 4, the first trunk is using the default port of 5060 and therefore only needs the user to distinguish incoming calls.&lt;br /&gt;
&lt;br /&gt;
[[File:PBX RR Trunk.PNG]]&lt;br /&gt;
&lt;br /&gt;
Outgoing calls will now be sent to port 5060 on the Grandstream to use lines 1-3, calls sent to 5160 will use line 4.  Incoming calls on lines 1-3 will hit the PBX with the user of 0001 which will select the first trunk, incoming calls on line 4 will hit the PBX with user 0002 selecting the 2nd trunk.&lt;br /&gt;
&lt;br /&gt;
== Remote Grandstream Configuration ==&lt;br /&gt;
&lt;br /&gt;
In order to get a Grandstream 410x working as a remote device, you must make a few changes from the standard, local configuration. Use this guide in conjunction with the standard guide.&lt;br /&gt;
&lt;br /&gt;
=== Grandstream Changes ===&lt;br /&gt;
&lt;br /&gt;
#On the Accounts=&amp;gt;Account X=&amp;gt;SIP Settings page, set SIP Registration to YES&lt;br /&gt;
#Under the User Accounts page, configure one channel with login credentials:&lt;br /&gt;
#Set the SIP server in the Profile to the PBX Public IP.&lt;br /&gt;
&lt;br /&gt;
[[File:Gxw410x remotesettings.PNG|File:Gxw410x remotesettings.PNG]]&lt;br /&gt;
&lt;br /&gt;
=== PBX SIP Provider Changes&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
#Instead of entering an IP address for Host, set to “dynamic”&lt;br /&gt;
#Set Register and Authentication to YES&lt;br /&gt;
#Set Username and Name to the value entered as SIP User ID and Authenticate ID in the grandstream&lt;br /&gt;
#Set Secret to the value entered for Authen Password in the grandstream&lt;br /&gt;
&lt;br /&gt;
Note: If the grandstream is not the only device at the remote site, then all remote SIP devices will need to have Can Reinvite set to YES.&lt;br /&gt;
&lt;br /&gt;
== Troubleshooting ==&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Outbound Dialing ===&lt;br /&gt;
&lt;br /&gt;
If you having trouble dialing outbound make the following changes on the FXO Lines page:&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;1. Tweak the Disconnect Threshold from 100 to 300ms.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;2. Tweak the Minimum Delay Before Dialing Out from 500 to 750ms.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Call Quality ===&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;If you are having call quality issues try the following changes on the Channels page:&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;1.Set Silence Suppression from YES to NO.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;2. Set Echo Cancellation from YES to NO&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Call Volume ===&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;If you are having issue with call volume try the following changes on the Channels page:&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;1.Increase/Decrease Tx to PSTN Audio Gain by increments of 3 for issues with external party volume.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;2. Increase/Decrease Rx from PSTN Audio Gain by increments of 3 for issues with internal party volume&lt;br /&gt;
&lt;br /&gt;
&amp;lt;span style=&amp;quot;font-size:large&amp;quot;&amp;gt;'''&amp;amp;nbsp;Procedure—Troubleshooting—Call Buzzing Noise'''&amp;lt;/span&amp;gt;&lt;br /&gt;
&lt;br /&gt;
If you are having issues with a buzz heard prior to a Menu prompt; try upgrading the Grandstream firmware:&lt;br /&gt;
&lt;br /&gt;
1. Navigate to Advanced Settings page&lt;br /&gt;
&lt;br /&gt;
2. Ensure HTTP is selected for the method to upgrade&lt;br /&gt;
&lt;br /&gt;
3. Set Firmware Server Path: to firmware.grandstream.com&lt;br /&gt;
&lt;br /&gt;
4. Set Automatic Upgrade to YES&lt;br /&gt;
&lt;br /&gt;
5. Set Allow DHCP Option 66 to override server to No Click Update at the bottom of the page&lt;br /&gt;
&lt;br /&gt;
6. Click Reboot The upgrade may take as long as 20min when done through the internet, so allow plenty of time for this. While upgrading the LED will blink. When the LED returns to normal, the device has completed its upgrade&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Restore Factory Default ===&lt;br /&gt;
&lt;br /&gt;
To Restore Factory Defaults:&lt;br /&gt;
&lt;br /&gt;
1. While powered up, hold the recessed Reset button in for 7+ seconds&lt;br /&gt;
&lt;br /&gt;
'''&amp;lt;span style=&amp;quot;font-size:large&amp;quot;&amp;gt;[[File:Gstream6.JPG|File:Gstream6.JPG]]&amp;amp;nbsp;&amp;lt;/span&amp;gt;'''&lt;br /&gt;
&lt;br /&gt;
'''&amp;lt;span style=&amp;quot;font-size:large&amp;quot;&amp;gt;Procedure Setup of QOS&amp;lt;/span&amp;gt;'''&lt;br /&gt;
&lt;br /&gt;
1. Access the IP of the Grandstream.&lt;br /&gt;
&lt;br /&gt;
2. Navigate in the Grandstream Web Admin to Networks – Advanced Settings&lt;br /&gt;
&lt;br /&gt;
3. Set the Layer 3 Diff Serv Value to 24 Which is CS3 or in Binary 011000&lt;br /&gt;
&lt;br /&gt;
4. Submit your changes and reboot&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;{{:Grandstream_FXO_FAQ}}&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=Grandstream_410X_Setup&amp;diff=5129</id>
		<title>Grandstream 410X Setup</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=Grandstream_410X_Setup&amp;diff=5129"/>
		<updated>2025-02-07T15:46:15Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Grandstream GXW 410X Gateway and HT841 FXO FXS Gateway */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;== Grandstream GXW 410X Gateway and HT841 FXO FXS Gateway ==&lt;br /&gt;
&lt;br /&gt;
If you are installing and HT841 gateway, you will program it the same as the GXW 410X below with 2 added settings.  &lt;br /&gt;
&lt;br /&gt;
# In the SIP provider page you will change the Port to Custom and make it 6062.&lt;br /&gt;
# In the HT841 you will wet the FXO Port settings for the Hunt Group to port 1. Active, and the rest of the ports to 1.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:HT841.png|frameless|600x600px]]&lt;br /&gt;
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[[File:HT841 ports.png|none|thumb|600x600px]]&lt;br /&gt;
[[File:Ht881.png|left|frameless|602x602px]]&lt;br /&gt;
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[[File:Ht881-2.png|left|thumb|600x600px]]&lt;br /&gt;
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Description&lt;br /&gt;
&lt;br /&gt;
The following configuration guide is for use setting up the Grandstream GXW410X gateways with analog circuits.&lt;br /&gt;
&lt;br /&gt;
NOTE: This guide is based upon the most current software from Grandstream. However if you have older firmware this is a link to the older guide which documents and covers the older firmware. &amp;amp;nbsp;The 4104 and 4108 use the exact same firmware as each other and have the same interface.&lt;br /&gt;
&lt;br /&gt;
[http://wiki.ipitomy.com/images/5/57/Tech_Bulletin_2011-008_Grandstream_Configuration_Guide.pdf http://wiki.ipitomy.com/images/5/57/Tech_Bulletin_2011-008_Grandstream_Configuration_Guide.pdf]&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream1.JPG|File:Gstream1.JPG]]&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Connections ===&lt;br /&gt;
&lt;br /&gt;
1. Connect each analog circuit to the desired FXO port, as above&lt;br /&gt;
&lt;br /&gt;
2. Connect the Grandstream WAN port to an available LAN port of the network switch/router being used on site.&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream2.JPG|File:Gstream2.JPG]]&lt;br /&gt;
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=== Procedure—Power Up and Login ===&lt;br /&gt;
&lt;br /&gt;
1. Power up the unit and identify its assigned IP address. (Typically assigned from the DCHP server of the host router.)&lt;br /&gt;
&lt;br /&gt;
2. Use your browser to access the Grandstream by inputting the IP Address assigned to it. The IP Address assigned by your router via DHCP can be discovered several ways – the easiest of which is likely by accessing the router’s connected devices page and finding it listed there.&lt;br /&gt;
&lt;br /&gt;
3. When the Grandstream page is accessed, input the password (admin at default) and navigate to the pages below making the changes as defined.&lt;br /&gt;
&lt;br /&gt;
=== IMPORTANT: Procedure Line Setup&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
On first login to the Grandstream, you will be brought to the following page.&lt;br /&gt;
&lt;br /&gt;
[[File:GrandstreamCPT.jpg|File:GrandstreamCPT.jpg]]&lt;br /&gt;
&lt;br /&gt;
In ALL new installs once you have the device on the Network and lines connected, you should run each of these tests. The first is for Call quality and is MOST important. As this test matches line impedance which if it is in variance can cause echo and poor calls. The second allows the device to learn Call Progress Tones (CPT) and is also very important so that the Unit can learn disconnect and busy tones on the lines. NOTE: The Grandstream can learn a Disconnect Supervision Current drop and a busy tone for disconnect. Lastly is the test for CID. This is also important and additionally can detect CID presence on the line. For proper performance of the Grandstream Device these tests SHOULD NOT be skipped.&lt;br /&gt;
&lt;br /&gt;
NOTE: These tests should be performed at the site and with the lines that the unit is intended to be in service upon.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuration&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
1. Navigate to the Networks Basic Setting page and input the data as in the example.&lt;br /&gt;
&lt;br /&gt;
[[File:SIP3.jpg|File:SIP3.jpg]]&amp;lt;br/&amp;gt;2. Assign a Static IP Address. The device must be found by the IPBX regardless of incidental changes and network adjustments. For this reason its best to change the IP Address to Static and assign an address that is out of range of those assigned for DCHP subscription. (E.g. if the router will assign DCHP Addresses from 192.168.1.1 ~ 192.168.1.50 you should select an IP Address out of this range ...192.168.1.200 would work unless it is being used elsewhere.)&amp;lt;br/&amp;gt;3. Use the other information provided by the DCHP assignment process in the remaining data fields; Subnet is usually 255.255.255.0. The Default Router Address must be that of the router the same one that assigns DCHP IP Addresses. DNS should also be the router since it will direct traffic.&amp;lt;br/&amp;gt;4. Click the save button. This saves information on this page before moving on.&amp;lt;br/&amp;gt;5. Navigate to the FXO Lines page. Then Select Dialing on the left.&amp;lt;br/&amp;gt;6. Change the Stage Method(1/2)&amp;lt;br/&amp;gt;Ch1-4:1;&lt;br /&gt;
&lt;br /&gt;
[[File:Gxw410x stagemethod.PNG|File:Gxw410x stagemethod.PNG]]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;7. Navigate to Settings and then select Channels Settings on the left. Program DTMF Methods to ch1-4:2;.  Program the Unconditional Call Forward to VOIP: to include the DID (Direct Inward Dial) number(s) that are to be routed.This routing is accomplished by Profiles 1, 2 &amp;amp; 3. Usually only one is necessary.This data field is the routing of the calls that are received on this FXO circuit (It is recomended to use our example on this line which is&amp;lt;br/&amp;gt;ch1:1111;ch2:2222;ch3:3333;ch4:4444;&amp;lt;br/&amp;gt;This will create &amp;quot;virtual DIDs&amp;quot; of 1111, 2222, etc, that allow you to enter DIDs in the PBX under the SIP Provider so you can route calls to different locations based on what line they came in on.&amp;amp;nbsp; We have programmed the Grandstream to route calls that have been received to the SIP Server using Profile 1.&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream3.JPG|File:Gstream3.JPG]]&lt;br /&gt;
&lt;br /&gt;
8.Click the save button.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuration Profile ===&lt;br /&gt;
&lt;br /&gt;
[[File:SIP.jpg|File:SIP.jpg]]&lt;br /&gt;
&lt;br /&gt;
1. Navigate to the Accounts=&amp;gt;Account 1=&amp;gt;General Settings page.&amp;lt;br/&amp;gt;2. Here the SIP Server must be programmed. Set this to be the IP Address of the IPBX. In our example, the address is 192.168.2.153.&amp;lt;br/&amp;gt;3. Also be sure that the SIP Registration field is set to No.&amp;lt;br/&amp;gt;4.Click the save button.&amp;lt;br/&amp;gt;5.The Grandstream configuration is now complete. However if you wish to make changes to the dial plan allowed digits you must also program that information.&amp;lt;br/&amp;gt;Note: At default the Grandstream allows only digits 0-9 to be sent to the connected PSTN circuit. If you want to use PSTN features like call forward you will need to change the Outgoing Dial Plan field to;&amp;lt;br/&amp;gt;{x+ | [x*]+}&lt;br /&gt;
&lt;br /&gt;
[[File:Sip2.jpg|File:Sip2.jpg]]&amp;lt;br/&amp;gt;to be able to send a * to outside IVR's&amp;lt;br/&amp;gt;6. When programming is complete in the Grandstream, click submit will commit the changes saved thus far to memory and make them operational. Continue to Procedure Configuring the IPitomy IP PBX&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream5.JPG|File:Gstream5.JPG]]&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuring the IPitomy IP PBX for Grandstream GXW 410x ===&lt;br /&gt;
&lt;br /&gt;
1.In the IPitomy IPBX set the fields as you see them below... using the Static IP Address assigned to the Procedure—Configuration step previously completed.. (In our example we assigned the Grandstream an IP Address of 192.168.2.9. This becomes the ―Host)&lt;br /&gt;
&lt;br /&gt;
2. Click “Save Changes”&lt;br /&gt;
&lt;br /&gt;
3. Then click on theApply Changes (upper right) to make these settings operational in the Ipitomy IPBX.&lt;br /&gt;
&lt;br /&gt;
4.Test the operation. Make a call into each of the Grandstream ports that have circuits and assure that they are being routed as defined in Call Routing—Incoming.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;5.Test the operation. Make a call at an Ipitomy extension using a calling pattern as defined in Call Routing Outgoing to assure that the call that should be placed over the Grandstream ports are placed.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuring Optional Outbound Routing Methods ===&lt;br /&gt;
&lt;br /&gt;
The Grandstream allows for two methods of outbound dialing at the same time.&lt;br /&gt;
&lt;br /&gt;
Round-Robin (Linear Hunt)&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;If you add the trunk to an outbound route in the PBX, the Grandstream will follow the Round-Robin rules, which are found on the Channels page. By default it will start at Line&lt;br /&gt;
&lt;br /&gt;
1, and move on until it finds an available channel to dial outbound.If you add the trunk to an outbound route and configure Prefix Digits to 99X where X is the port on the Grandstream you wish to use when placing this call (e.g. 991 is line 1). By doing this you can configure the calls to route out a particular line, or a different order if you add the trunk multiple times and prefix accordingly (992, 994, 991, 993, etc). The code of 99 can be changed on the Channels page in the Grandstream.&lt;br /&gt;
&lt;br /&gt;
1. Navigate to the Channels page&lt;br /&gt;
&lt;br /&gt;
2. Go to the Port Scheduling Schema (Voip-&amp;gt;PSTN) section and input the code or codes that you wish (99x)&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configure Multiple Line Groups on Single Grandstream ===&lt;br /&gt;
&lt;br /&gt;
'''1 -''' In order to put the lines into separate outbound groups you will need to configure each round robin group under FXO Lines =&amp;gt; Dialing.  In this example lines 1-3 are in the first group and line 4 is standalone.&lt;br /&gt;
&lt;br /&gt;
[[File:GS RR Separate.PNG|File:GS RR Separate.PNG]]&lt;br /&gt;
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&lt;br /&gt;
'''2 -''' After setting up the round robin groups you now have to configure the local SIP port for the groups.  In this example channels 1-3 are using 5060 and incrementing by 2 (this is default behavior for multiple channels in a group, any call sent to 5060 will use the first available line in the group), channel 4 is alone listening on port 5160.&lt;br /&gt;
&lt;br /&gt;
[[File:GS RR Ports.PNG]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''3 -''' After configuring the local SIP port you now need to configure the account SIP User ID, this will allow the PBX to differentiate between each line group on incoming calls.  Lines 1-3 will give a user ID of 0001, and line 4 will give a user ID of 0002.&lt;br /&gt;
&lt;br /&gt;
[[File:GS RR Account.PNG]]&lt;br /&gt;
&lt;br /&gt;
'''4 -''' On the PBX side we will need to create two trunks, both pointing to the same IP address of the Grandstream, but each will have a different user.  '''Be absolutely certain you configure the trunk with Insecure = No''', this setting will tell the PBX to consider the user when taking the call.  Notice that on the 2nd trunk we've added the correct port number for line 4 and the correct User associated with line 4.  These two settings are what make the 2nd trunk use line 4, the first trunk is using the default port of 5060 and therefore only needs the user to distinguish incoming calls.&lt;br /&gt;
&lt;br /&gt;
[[File:PBX RR Trunk.PNG]]&lt;br /&gt;
&lt;br /&gt;
Outgoing calls will now be sent to port 5060 on the Grandstream to use lines 1-3, calls sent to 5160 will use line 4.  Incoming calls on lines 1-3 will hit the PBX with the user of 0001 which will select the first trunk, incoming calls on line 4 will hit the PBX with user 0002 selecting the 2nd trunk.&lt;br /&gt;
&lt;br /&gt;
== Remote Grandstream Configuration ==&lt;br /&gt;
&lt;br /&gt;
In order to get a Grandstream 410x working as a remote device, you must make a few changes from the standard, local configuration. Use this guide in conjunction with the standard guide.&lt;br /&gt;
&lt;br /&gt;
=== Grandstream Changes ===&lt;br /&gt;
&lt;br /&gt;
#On the Accounts=&amp;gt;Account X=&amp;gt;SIP Settings page, set SIP Registration to YES&lt;br /&gt;
#Under the User Accounts page, configure one channel with login credentials:&lt;br /&gt;
#Set the SIP server in the Profile to the PBX Public IP.&lt;br /&gt;
&lt;br /&gt;
[[File:Gxw410x remotesettings.PNG|File:Gxw410x remotesettings.PNG]]&lt;br /&gt;
&lt;br /&gt;
=== PBX SIP Provider Changes&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
#Instead of entering an IP address for Host, set to “dynamic”&lt;br /&gt;
#Set Register and Authentication to YES&lt;br /&gt;
#Set Username and Name to the value entered as SIP User ID and Authenticate ID in the grandstream&lt;br /&gt;
#Set Secret to the value entered for Authen Password in the grandstream&lt;br /&gt;
&lt;br /&gt;
Note: If the grandstream is not the only device at the remote site, then all remote SIP devices will need to have Can Reinvite set to YES.&lt;br /&gt;
&lt;br /&gt;
== Troubleshooting ==&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Outbound Dialing ===&lt;br /&gt;
&lt;br /&gt;
If you having trouble dialing outbound make the following changes on the FXO Lines page:&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;1. Tweak the Disconnect Threshold from 100 to 300ms.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;2. Tweak the Minimum Delay Before Dialing Out from 500 to 750ms.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Call Quality ===&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;If you are having call quality issues try the following changes on the Channels page:&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;1.Set Silence Suppression from YES to NO.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;2. Set Echo Cancellation from YES to NO&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Call Volume ===&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;If you are having issue with call volume try the following changes on the Channels page:&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;1.Increase/Decrease Tx to PSTN Audio Gain by increments of 3 for issues with external party volume.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;2. Increase/Decrease Rx from PSTN Audio Gain by increments of 3 for issues with internal party volume&lt;br /&gt;
&lt;br /&gt;
&amp;lt;span style=&amp;quot;font-size:large&amp;quot;&amp;gt;'''&amp;amp;nbsp;Procedure—Troubleshooting—Call Buzzing Noise'''&amp;lt;/span&amp;gt;&lt;br /&gt;
&lt;br /&gt;
If you are having issues with a buzz heard prior to a Menu prompt; try upgrading the Grandstream firmware:&lt;br /&gt;
&lt;br /&gt;
1. Navigate to Advanced Settings page&lt;br /&gt;
&lt;br /&gt;
2. Ensure HTTP is selected for the method to upgrade&lt;br /&gt;
&lt;br /&gt;
3. Set Firmware Server Path: to firmware.grandstream.com&lt;br /&gt;
&lt;br /&gt;
4. Set Automatic Upgrade to YES&lt;br /&gt;
&lt;br /&gt;
5. Set Allow DHCP Option 66 to override server to No Click Update at the bottom of the page&lt;br /&gt;
&lt;br /&gt;
6. Click Reboot The upgrade may take as long as 20min when done through the internet, so allow plenty of time for this. While upgrading the LED will blink. When the LED returns to normal, the device has completed its upgrade&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Restore Factory Default ===&lt;br /&gt;
&lt;br /&gt;
To Restore Factory Defaults:&lt;br /&gt;
&lt;br /&gt;
1. While powered up, hold the recessed Reset button in for 7+ seconds&lt;br /&gt;
&lt;br /&gt;
'''&amp;lt;span style=&amp;quot;font-size:large&amp;quot;&amp;gt;[[File:Gstream6.JPG|File:Gstream6.JPG]]&amp;amp;nbsp;&amp;lt;/span&amp;gt;'''&lt;br /&gt;
&lt;br /&gt;
'''&amp;lt;span style=&amp;quot;font-size:large&amp;quot;&amp;gt;Procedure Setup of QOS&amp;lt;/span&amp;gt;'''&lt;br /&gt;
&lt;br /&gt;
1. Access the IP of the Grandstream.&lt;br /&gt;
&lt;br /&gt;
2. Navigate in the Grandstream Web Admin to Networks – Advanced Settings&lt;br /&gt;
&lt;br /&gt;
3. Set the Layer 3 Diff Serv Value to 24 Which is CS3 or in Binary 011000&lt;br /&gt;
&lt;br /&gt;
4. Submit your changes and reboot&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;{{:Grandstream_FXO_FAQ}}&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=Call_Recording&amp;diff=5128</id>
		<title>Call Recording</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=Call_Recording&amp;diff=5128"/>
		<updated>2025-01-31T21:30:09Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Manual Recordings:   */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;IMPORTANT: In many jurisdictions, it is a requirement to announce the call is being recorded.&amp;amp;nbsp; To that end, you can insert this prompt into either a menu that fails over to the recorded ring group, or if you have ACD, as an intro announcement on the acd group being recorded itself:&amp;amp;nbsp;&amp;amp;nbsp;[https://www.dropbox.com/s/nplru2ea5iqtxgn/this-call-may-be-monitored-or-recorded.wav?dl=0 Call May Be Recorded audio file]&lt;br /&gt;
&lt;br /&gt;
This feature allows recording of inbound and outbound calls. Extensions can be put into groups which makes it easier to keep track of calls. There can be jobs that only record inbound calls, outbound calls, or both.&lt;br /&gt;
&lt;br /&gt;
==Recording Methods==&lt;br /&gt;
&lt;br /&gt;
=== Manual Recordings: &amp;amp;nbsp; ===&lt;br /&gt;
*Initiated by agent, or Extension User while on call by dialing *# or pressing a programmed key (a BLF set to speed dial) on the phone.&amp;amp;nbsp;&lt;br /&gt;
**You will need to ensure that the Call Recording is enabled in the proper places for this to work:&lt;br /&gt;
***Trunks&lt;br /&gt;
***Ring Groups&lt;br /&gt;
***Extension initiating the recording&lt;br /&gt;
***Extension being recorded&lt;br /&gt;
***Extension Advanced settings must have &amp;quot;Is Operator&amp;quot; checked.&lt;br /&gt;
'''''ATTENTION!'''''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;Recordings created in this manner can be found in the user's Smart Personal Console (the &amp;quot;user&amp;quot; web browser log in with the extension PIN or VM password). Click your Voicemail then click on the &amp;quot;Work&amp;quot; folder.&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
Or they can be found in Agent Recordings ('''''IF''''' they are recorded by an agent taking a queue call).&lt;br /&gt;
&lt;br /&gt;
=== Initiated by QManager ===&lt;br /&gt;
**These recordings are available through View Recordings in QManager&lt;br /&gt;
=== Third Party Recording ===&lt;br /&gt;
*IPitomy has implemented integration with Trivium to support their call recording application.&lt;br /&gt;
=== IPitomy Call Recording Application ===&lt;br /&gt;
*Licensed IPitomy Feature that records all calls on designated destinations.&lt;br /&gt;
**NOTE: This cannot record Conference Rooms&lt;br /&gt;
&lt;br /&gt;
NOTE: When using the IPitomy Call Recording feature, it is advised to configure an FTP to push those recordings to so they can be keep for a long time, and to set the recording job to purge after an FTP push so that the file size does not get too large. [[File:Callrecording.png|center|Callrecording.png]]&lt;br /&gt;
&lt;br /&gt;
== Create New Recording Job ==&lt;br /&gt;
&lt;br /&gt;
#Navigate to the Call Recording Page under the Applications tab.&lt;br /&gt;
#Click Add Recording&lt;br /&gt;
#Enter the name of the recording group.&lt;br /&gt;
#Add Ring Groups to the recording job that are applicable to the group that you want to be recorded.&lt;br /&gt;
#Add Available Routes that you would like to be recorded.&lt;br /&gt;
#Click on the [[File:Savechanges.png]] button.&lt;br /&gt;
&amp;lt;p style=&amp;quot;text-align: center&amp;quot;&amp;gt;[[File:RecordingJob.jpg|File:RecordingJob.jpg]]&amp;lt;/p&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{| class=&amp;quot;wikitable&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
! Sections/Fields&lt;br /&gt;
! Description&lt;br /&gt;
|-&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | '''Recording Settings'''&lt;br /&gt;
|-&lt;br /&gt;
| Name&lt;br /&gt;
| Enter the name of the Recording Job&lt;br /&gt;
|-&lt;br /&gt;
| Enable&lt;br /&gt;
| Option to enable or disable call recording&lt;br /&gt;
|-&lt;br /&gt;
| Maximum number of recordings&lt;br /&gt;
| Enter the maximum amount of calls you would like to have recorded. If the maximum number has been reached, then the old recordings will be deleted to make room for new ones.&lt;br /&gt;
|-&lt;br /&gt;
| Maximum age of recordings&lt;br /&gt;
| Enter the maximum age of recordings. This will delete recordings after a certain period of time. This will clear out old recordings automatically to make room for new recordings.&lt;br /&gt;
|-&lt;br /&gt;
| Available Ring Groups&lt;br /&gt;
| These are the groups that can be selected to include in the recording job&lt;br /&gt;
|-&lt;br /&gt;
| Selected&lt;br /&gt;
| These are the groups that have been added to be included in the recording job. These are the extensions that you will find recordings for when checking this recording job&lt;br /&gt;
|-&lt;br /&gt;
| Available Routes&lt;br /&gt;
| Routes that can be selected to be included in the recording job. When one of these routes is dialed by any extensions on the system, the call will be recorded.&lt;br /&gt;
|-&lt;br /&gt;
| Selected&lt;br /&gt;
| These are the routes that have been selected to be recorded when dialed by the selected ring groups.&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
=== Viewing Recording Jobs ===&lt;br /&gt;
&lt;br /&gt;
[[File:Managerecordingfiles.png|center|Managerecordingfiles.png]]&lt;br /&gt;
&lt;br /&gt;
To View the recording jobs and manage or listen to the recorded calls:&lt;br /&gt;
&lt;br /&gt;
#Navigate to the Call Recording page under the Applications tab.&lt;br /&gt;
#Click the [[File:Viewjobs.png]] icon to view the recording files for the group you wish to view.&lt;br /&gt;
#To download/listen to the recording files, click the [[File:Downloadbutton.jpg]] button.&lt;br /&gt;
&lt;br /&gt;
=== Archived Recordings ===&lt;br /&gt;
&lt;br /&gt;
After a day, the PBX will archive the recordings from the previous day. These are put into a .zip folder and can be viewed on the PBX or downloaded. If you download them, there will be an index.html file contained within the .zip that gives information about each recording (Recording Target, Filename, Size, Date, Source &amp;amp; Destination) so you can find the recording you are looking for. You can also then click the link and it will play the file you want.&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=Call_Recording&amp;diff=5127</id>
		<title>Call Recording</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=Call_Recording&amp;diff=5127"/>
		<updated>2024-12-13T17:51:15Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;IMPORTANT: In many jurisdictions, it is a requirement to announce the call is being recorded.&amp;amp;nbsp; To that end, you can insert this prompt into either a menu that fails over to the recorded ring group, or if you have ACD, as an intro announcement on the acd group being recorded itself:&amp;amp;nbsp;&amp;amp;nbsp;[https://www.dropbox.com/s/nplru2ea5iqtxgn/this-call-may-be-monitored-or-recorded.wav?dl=0 Call May Be Recorded audio file]&lt;br /&gt;
&lt;br /&gt;
This feature allows recording of inbound and outbound calls. Extensions can be put into groups which makes it easier to keep track of calls. There can be jobs that only record inbound calls, outbound calls, or both.&lt;br /&gt;
&lt;br /&gt;
==Recording Methods==&lt;br /&gt;
&lt;br /&gt;
=== Manual Recordings: &amp;amp;nbsp; ===&lt;br /&gt;
*Initiated by agent, or Extension User while on call by dialing *#*# or pressing a programmed key (a BLF set to speed dial) on the phone.&amp;amp;nbsp;&lt;br /&gt;
**You will need to ensure that the Call Recording is enabled in the proper places for this to work:&lt;br /&gt;
***Trunks&lt;br /&gt;
***Ring Groups&lt;br /&gt;
***Extension initiating the recording&lt;br /&gt;
***Extension being recorded&lt;br /&gt;
***Extension Advanced settings must have &amp;quot;Is Operator&amp;quot; checked.&lt;br /&gt;
'''''ATTENTION!'''''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;big&amp;gt;Recordings created in this manner can be found in the user's Smart Personal Console (the &amp;quot;user&amp;quot; web browser log in with the extension PIN or VM password). Click your Voicemail then click on the &amp;quot;Work&amp;quot; folder.&amp;lt;/big&amp;gt;&lt;br /&gt;
&lt;br /&gt;
Or they can be found in Agent Recordings ('''''IF''''' they are recorded by an agent taking a queue call).&lt;br /&gt;
&lt;br /&gt;
=== Initiated by QManager ===&lt;br /&gt;
**These recordings are available through View Recordings in QManager&lt;br /&gt;
=== Third Party Recording ===&lt;br /&gt;
*IPitomy has implemented integration with Trivium to support their call recording application.&lt;br /&gt;
=== IPitomy Call Recording Application ===&lt;br /&gt;
*Licensed IPitomy Feature that records all calls on designated destinations.&lt;br /&gt;
**NOTE: This cannot record Conference Rooms&lt;br /&gt;
&lt;br /&gt;
NOTE: When using the IPitomy Call Recording feature, it is advised to configure an FTP to push those recordings to so they can be keep for a long time, and to set the recording job to purge after an FTP push so that the file size does not get too large. [[File:Callrecording.png|center|Callrecording.png]]&lt;br /&gt;
&lt;br /&gt;
== Create New Recording Job ==&lt;br /&gt;
&lt;br /&gt;
#Navigate to the Call Recording Page under the Applications tab.&lt;br /&gt;
#Click Add Recording&lt;br /&gt;
#Enter the name of the recording group.&lt;br /&gt;
#Add Ring Groups to the recording job that are applicable to the group that you want to be recorded.&lt;br /&gt;
#Add Available Routes that you would like to be recorded.&lt;br /&gt;
#Click on the [[File:Savechanges.png]] button.&lt;br /&gt;
&amp;lt;p style=&amp;quot;text-align: center&amp;quot;&amp;gt;[[File:RecordingJob.jpg|File:RecordingJob.jpg]]&amp;lt;/p&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{| class=&amp;quot;wikitable&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
! Sections/Fields&lt;br /&gt;
! Description&lt;br /&gt;
|-&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | '''Recording Settings'''&lt;br /&gt;
|-&lt;br /&gt;
| Name&lt;br /&gt;
| Enter the name of the Recording Job&lt;br /&gt;
|-&lt;br /&gt;
| Enable&lt;br /&gt;
| Option to enable or disable call recording&lt;br /&gt;
|-&lt;br /&gt;
| Maximum number of recordings&lt;br /&gt;
| Enter the maximum amount of calls you would like to have recorded. If the maximum number has been reached, then the old recordings will be deleted to make room for new ones.&lt;br /&gt;
|-&lt;br /&gt;
| Maximum age of recordings&lt;br /&gt;
| Enter the maximum age of recordings. This will delete recordings after a certain period of time. This will clear out old recordings automatically to make room for new recordings.&lt;br /&gt;
|-&lt;br /&gt;
| Available Ring Groups&lt;br /&gt;
| These are the groups that can be selected to include in the recording job&lt;br /&gt;
|-&lt;br /&gt;
| Selected&lt;br /&gt;
| These are the groups that have been added to be included in the recording job. These are the extensions that you will find recordings for when checking this recording job&lt;br /&gt;
|-&lt;br /&gt;
| Available Routes&lt;br /&gt;
| Routes that can be selected to be included in the recording job. When one of these routes is dialed by any extensions on the system, the call will be recorded.&lt;br /&gt;
|-&lt;br /&gt;
| Selected&lt;br /&gt;
| These are the routes that have been selected to be recorded when dialed by the selected ring groups.&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
=== Viewing Recording Jobs ===&lt;br /&gt;
&lt;br /&gt;
[[File:Managerecordingfiles.png|center|Managerecordingfiles.png]]&lt;br /&gt;
&lt;br /&gt;
To View the recording jobs and manage or listen to the recorded calls:&lt;br /&gt;
&lt;br /&gt;
#Navigate to the Call Recording page under the Applications tab.&lt;br /&gt;
#Click the [[File:Viewjobs.png]] icon to view the recording files for the group you wish to view.&lt;br /&gt;
#To download/listen to the recording files, click the [[File:Downloadbutton.jpg]] button.&lt;br /&gt;
&lt;br /&gt;
=== Archived Recordings ===&lt;br /&gt;
&lt;br /&gt;
After a day, the PBX will archive the recordings from the previous day. These are put into a .zip folder and can be viewed on the PBX or downloaded. If you download them, there will be an index.html file contained within the .zip that gives information about each recording (Recording Target, Filename, Size, Date, Source &amp;amp; Destination) so you can find the recording you are looking for. You can also then click the link and it will play the file you want.&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPitomy_Communicator&amp;diff=5126</id>
		<title>IPitomy Communicator</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPitomy_Communicator&amp;diff=5126"/>
		<updated>2024-12-10T15:30:54Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Note: This Guide will assist the user in Installing and Using the IPitomy Communicator Softphone.&lt;br /&gt;
&lt;br /&gt;
We have two different soft phone installs. The “demo” version does not include the user list which shows all of the presence of all of the users on the system. This is intended for trial versions or for users who are on a shared instance of IPitomy in the Cloud. For obvious reasons, on a shared instance, it would not be good to display the presence of all the other users!&lt;br /&gt;
&lt;br /&gt;
The full commercial version is called IPitomy Communicator.&lt;br /&gt;
&lt;br /&gt;
This requires an available IPitomy Extension License as well as an IPitomy Soft Phone License. This does not have auto configuration, but is super simple to set up. When installing it, select IP620 as the device type for the extension so it will use an IPitomy license instead of an open license.&lt;br /&gt;
&lt;br /&gt;
This also requires the generation of a single API Key. If your PBX already has one generated for the IPitomy Communicator, you don't need to do anything else. If not, simply navigate to Applications=&amp;gt;API Key, name it Softphone [one word, capital S], and click Create.&amp;amp;nbsp; Be sure to enable it by checking the box labeled &amp;quot;Key Enable&amp;quot;, then save and apply changes.&lt;br /&gt;
&lt;br /&gt;
Links for SoftPhone:&lt;br /&gt;
[[File:Screenshot for communicator.png|alt=Screenshot for communicator|thumb|518x518px|Screenshot for communicator]]&lt;br /&gt;
[[File:Screenshot for communicator 2.png|thumb|519x519px|Screenshot for communicator 2]]&lt;br /&gt;
Demo version: [http://relay.ipitomy.com/softcall_demo/publish.htm http://relay.ipitomy.com/softcall_demo/publish.htm]&lt;br /&gt;
&lt;br /&gt;
Full version – IPitomy Communicator: [https://www.dropbox.com/s/6uhj12mwh8k7diu/CommunicatorSetup.zip?dl=0 &amp;lt;nowiki&amp;gt;[Here]&amp;lt;/nowiki&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''''&amp;lt;big&amp;gt;&amp;lt;u&amp;gt;IMPORTANT&amp;lt;/u&amp;gt;&amp;lt;/big&amp;gt;'''''&lt;br /&gt;
&lt;br /&gt;
To avoid an error, you must also install ''Visual C++ Redistributable for Visual Studio 2012 Update 4''.&lt;br /&gt;
&lt;br /&gt;
The file is here:  '''''&amp;lt;big&amp;gt;https://www.microsoft.com/en-us/download/details.aspx?id=30679&amp;lt;/big&amp;gt;'''''&lt;br /&gt;
&lt;br /&gt;
Scroll down just a little bit, click the download button, then select the x86 version (yes, even if you're running 64 bit Windows).&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== '''Ipitomy Softphone'''&amp;lt;br/&amp;gt; ==&lt;br /&gt;
&lt;br /&gt;
'''Installation'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Installation Video'''&lt;br /&gt;
&lt;br /&gt;
[https://www.youtube.com/watch?v=p5gwURMp4eo &amp;lt;b&amp;gt;https://www.youtube.com/watch?v=p5gwURMp4eo&amp;lt;/b&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Park Button Video'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[https://www.youtube.com/watch?v=dPCVXiR6kf8 &amp;lt;b&amp;gt;https://www.youtube.com/watch?v=dPCVXiR6kf8&amp;lt;/b&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Install the Softphone, Follow the URL above and click the Installer link. Installation will begin at that time. You should have adminstrative rights to the PC you are installing to prior to installation. Accept the default location for install unless you have some other directory you wish to install to.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Setup'''&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Once isntalled open the program by clicking the program Icon. This should now be located on your Desktop as well as in the Program directory. The Program will open bringing up the user interface screen.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone3.jpg|File:SoftPhone3.jpg]][[File:SoftPhone4.jpg|File:SoftPhone4.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Add an account Press the Settings Icon which is the little Gear Icon next to the Ipitomy Logo Bar. Select Sip Accounts and Add an Account&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone5.jpg|File:SoftPhone5.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Select the Add Button and Enter the Following:&lt;br /&gt;
&lt;br /&gt;
Display Name = Name of user&lt;br /&gt;
&lt;br /&gt;
User Name = Extension Number&lt;br /&gt;
&lt;br /&gt;
User Password = Voicemail Password&lt;br /&gt;
&lt;br /&gt;
SIP Password = SIP Password for Extension&lt;br /&gt;
&lt;br /&gt;
SIP Server = IP of PBX (Local IP is OK for a Local Extension only, however for a remote phone this would need to the Public IP of the Network the PBX resides upon)&lt;br /&gt;
&lt;br /&gt;
Integration Port = Leave blank and the software uses 80 to verify its license with the PBX.&amp;amp;nbsp; If the softphone is remote and an external port other than 80 is forwarded to the PBX, enter that as the Integration Port (eg, 8080 external forwarded to 80 internal)&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
All other information can be left at Default.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Communicator-SIP Acc.jpg|File:Communicator-SIP Acc.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Be Sure to Click Enabled when you are returned to the account screen on the account you just created so it is made the live account&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone8.jpg|File:SoftPhone8.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Click close and you should be returned to the User interface and your account should register to the PBX.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone9.jpg|File:SoftPhone9.jpg]][[File:SoftPhone10.jpg|File:SoftPhone10.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Your Softphone is now ready to be used.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Import Extensions from the PBX to Contacts.'''&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Import all Extension in the PBX into the Contact list select User List - Manage Users - Ipitomy and then Refresh&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone11.jpg|File:SoftPhone11.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Select the PBX from the Host list&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone14.jpg|File:SoftPhone14.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Click OK. The Softphone will now pull down an Extension list from the PBX which you configured.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone13.jpg|File:SoftPhone13.jpg]]&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPitomy_Communicator&amp;diff=5125</id>
		<title>IPitomy Communicator</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPitomy_Communicator&amp;diff=5125"/>
		<updated>2024-12-10T15:30:15Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Note: This Guide will assist the user in Installing and Using the IPitomy Communicator Softphone.&lt;br /&gt;
&lt;br /&gt;
We have two different soft phone installs. The “demo” version does not include the user list which shows all of the presence of all of the users on the system. This is intended for trial versions or for users who are on a shared instance of IPitomy in the Cloud. For obvious reasons, on a shared instance, it would not be good to display the presence of all the other users!&lt;br /&gt;
&lt;br /&gt;
The full commercial version is called IPitomy Communicator.&lt;br /&gt;
&lt;br /&gt;
This requires an available IPitomy Extension License as well as an IPitomy Soft Phone License. This does not have auto configuration, but is super simple to set up. When installing it, select IP620 as the device type for the extension so it will use an IPitomy license instead of an open license.&lt;br /&gt;
&lt;br /&gt;
This also requires the generation of a single API Key. If your PBX already has one generated for the IPitomy Communicator, you don't need to do anything else. If not, simply navigate to Applications=&amp;gt;API Key, name it Softphone [one word, capital S], and click Create.&amp;amp;nbsp; Be sure to enable it by checking the box labeled &amp;quot;Key Enable&amp;quot;, then save and apply changes.&lt;br /&gt;
&lt;br /&gt;
Links for SoftPhone:&lt;br /&gt;
[[File:Screenshot for communicator.png|alt=Screenshot for communicator|thumb|518x518px|Screenshot for communicator]]&lt;br /&gt;
[[File:Screenshot for communicator 2.png|thumb|519x519px|Screenshot for communicator 2]]&lt;br /&gt;
Demo version: [http://relay.ipitomy.com/softcall_demo/publish.htm http://relay.ipitomy.com/softcall_demo/publish.htm]&lt;br /&gt;
&lt;br /&gt;
Full version – IPitomy Communicator: [https://www.dropbox.com/s/6uhj12mwh8k7diu/CommunicatorSetup.zip?dl=0 &amp;lt;nowiki&amp;gt;[Here]&amp;lt;/nowiki&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''''&amp;lt;big&amp;gt;&amp;lt;u&amp;gt;IMPORTANT&amp;lt;/u&amp;gt;&amp;lt;/big&amp;gt;'''''&lt;br /&gt;
&lt;br /&gt;
To avoid an error, you must also install ''Visual C++ Redistributable for Visual Studio 2012 Update 4''.&lt;br /&gt;
&lt;br /&gt;
The file is here:  '''''&amp;lt;big&amp;gt;https://www.microsoft.com/en-us/download/details.aspx?id=30679&amp;lt;/big&amp;gt;'''''&lt;br /&gt;
&lt;br /&gt;
Scroll down just a little bit, click the download button, then select the x86 version (yes, even if you're running 64 bit Windows).&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== '''Ipitomy Softphone'''&amp;lt;br/&amp;gt; ==&lt;br /&gt;
&lt;br /&gt;
'''Installation'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Installation Video'''&lt;br /&gt;
&lt;br /&gt;
[https://www.youtube.com/watch?v=p5gwURMp4eo &amp;lt;b&amp;gt;https://www.youtube.com/watch?v=p5gwURMp4eo&amp;lt;/b&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Park Button Video'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[https://www.youtube.com/watch?v=dPCVXiR6kf8 &amp;lt;b&amp;gt;https://www.youtube.com/watch?v=dPCVXiR6kf8&amp;lt;/b&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Install the Softphone, Follow the URL above and click the Installer link. Installation will begin at that time. You should have adminstrative rights to the PC you are installing to prior to installation. Accept the default location for install unless you have some other directory you wish to install to.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Setup'''&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Once isntalled open the program by clicking the program Icon. This should now be located on your Desktop as well as in the Program directory. The Program will open bringing up the user interface screen.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone3.jpg|File:SoftPhone3.jpg]][[File:SoftPhone4.jpg|File:SoftPhone4.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Add an account Press the Settings Icon which is the little Gear Icon next to the Ipitomy Logo Bar. Select Sip Accounts and Add an Account&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone5.jpg|File:SoftPhone5.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Select the Add Button and Enter the Following:&lt;br /&gt;
&lt;br /&gt;
Display Name = Name of user&lt;br /&gt;
&lt;br /&gt;
User Name = Extension Number&lt;br /&gt;
&lt;br /&gt;
User Password = Voicemail Password&lt;br /&gt;
&lt;br /&gt;
SIP Password = SIP Password for Extension&lt;br /&gt;
&lt;br /&gt;
SIP Server = IP of PBX (Local IP is OK for a Local Extension only, however for a remote phone this would need to the Public IP of the Network the PBX resides upon)&lt;br /&gt;
&lt;br /&gt;
Integration Port = Leave blank and the software uses 80 to verify its license with the PBX.&amp;amp;nbsp; If the softphone is remote and an external port other than 80 is forwarded to the PBX, enter that as the Integration Port (eg, 8080 external forwarded to 80 internal)&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
All other information can be left at Default.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Communicator-SIP Acc.jpg|File:Communicator-SIP Acc.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Be Sure to Click Enabled when you are returned to the account screen on the account you just created so it is made the live account&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone8.jpg|File:SoftPhone8.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Click close and you should be returned to the User interface and your account should register to the PBX.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone9.jpg|File:SoftPhone9.jpg]][[File:SoftPhone10.jpg|File:SoftPhone10.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Your Softphone is now ready to be used.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Import Extensions from the PBX to Contacts.'''&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Import all Extension in the PBX into the Contact list select User List - Manage Users - Ipitomy and then Refresh&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone11.jpg|File:SoftPhone11.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Select the PBX from the Host list&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone14.jpg|File:SoftPhone14.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Click OK. The Softphone will now pull down an Extension list from the PBX which you configured.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone13.jpg|File:SoftPhone13.jpg]]&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPitomy_Communicator&amp;diff=5124</id>
		<title>IPitomy Communicator</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPitomy_Communicator&amp;diff=5124"/>
		<updated>2024-12-10T15:29:54Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Note: This Guide will assist the user in Installing and Using the IPitomy Communicator Softphone.&lt;br /&gt;
&lt;br /&gt;
We have two different soft phone installs. The “demo” version does not include the user list which shows all of the presence of all of the users on the system. This is intended for trial versions or for users who are on a shared instance of IPitomy in the Cloud. For obvious reasons, on a shared instance, it would not be good to display the presence of all the other users!&lt;br /&gt;
&lt;br /&gt;
The full commercial version is called IPitomy Communicator.&lt;br /&gt;
&lt;br /&gt;
This requires an available IPitomy Extension License as well as an IPitomy Soft Phone License. This does not have auto configuration, but is super simple to set up. When installing it, select IP620 as the device type for the extension so it will use an IPitomy license instead of an open license.&lt;br /&gt;
&lt;br /&gt;
This also requires the generation of a single API Key. If your PBX already has one generated for the IPitomy Communicator, you don't need to do anything else. If not, simply navigate to Applications=&amp;gt;API Key, name it Softphone [one word, capital S], and click Create.&amp;amp;nbsp; Be sure to enable it by checking the box labeled &amp;quot;Key Enable&amp;quot;, then save and apply changes.&lt;br /&gt;
&lt;br /&gt;
Links for SoftPhone:&lt;br /&gt;
[[File:Screenshot for communicator.png|alt=Screenshot for communicator|thumb|518x518px|Screenshot for communicator]]&lt;br /&gt;
[[File:Screenshot for communicator 2.png|thumb|519x519px|Screenshot for communicator 2]]&lt;br /&gt;
Demo version: [http://relay.ipitomy.com/softcall_demo/publish.htm http://relay.ipitomy.com/softcall_demo/publish.htm]&lt;br /&gt;
&lt;br /&gt;
Full version – IPitomy Communicator: [https://www.dropbox.com/s/6uhj12mwh8k7diu/CommunicatorSetup.zip?dl=0 &amp;lt;nowiki&amp;gt;[Here]&amp;lt;/nowiki&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''''&amp;lt;big&amp;gt;&amp;lt;u&amp;gt;IMPORTANT&amp;lt;/u&amp;gt;&amp;lt;/big&amp;gt;'''''&lt;br /&gt;
&lt;br /&gt;
To avoid an error, you must also install ''Visual C++ Redistributable for Visual Studio 2012 Update 4''.&lt;br /&gt;
&lt;br /&gt;
The file is here:  '''''&amp;lt;big&amp;gt;https://www.microsoft.com/en-us/download/details.aspx?id=30679&amp;lt;/big&amp;gt;'''''&lt;br /&gt;
&lt;br /&gt;
Scroll down just a little bit, click the download button, then select the x86 version (yes, even if you're running 64 bit Windows).&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== '''Ipitomy Softphone'''&amp;lt;br/&amp;gt; ==&lt;br /&gt;
&lt;br /&gt;
'''Installation'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Installation Video'''&lt;br /&gt;
&lt;br /&gt;
[https://www.youtube.com/watch?v=p5gwURMp4eo &amp;lt;b&amp;gt;https://www.youtube.com/watch?v=p5gwURMp4eo&amp;lt;/b&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Park Button Video'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[https://www.youtube.com/watch?v=dPCVXiR6kf8 &amp;lt;b&amp;gt;https://www.youtube.com/watch?v=dPCVXiR6kf8&amp;lt;/b&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Install the Softphone, Follow the URL above and click the Installer link. Installation will begin at that time. You should have adminstrative rights to the PC you are installing to prior to installation. Accept the default location for install unless you have some other directory you wish to install to.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Setup'''&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Once isntalled open the program by clicking the program Icon. This should now be located on your Desktop as well as in the Program directory. The Program will open bringing up the user interface screen.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone3.jpg|File:SoftPhone3.jpg]][[File:SoftPhone4.jpg|File:SoftPhone4.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Add an account Press the Settings Icon which is the little Gear Icon next to the Ipitomy Logo Bar. Select Sip Accounts and Add an Account&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone5.jpg|File:SoftPhone5.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Select the Add Button and Enter the Following:&lt;br /&gt;
&lt;br /&gt;
Display Name = Name of user&lt;br /&gt;
&lt;br /&gt;
User Name = Extension Number&lt;br /&gt;
&lt;br /&gt;
User Password = Voicemail Password&lt;br /&gt;
&lt;br /&gt;
SIP Password = SIP Password for Extension&lt;br /&gt;
&lt;br /&gt;
SIP Server = IP of PBX (Local IP is OK for a Local Extension only, however for a remote phone this would need to the Public IP of the Network the PBX resides upon)&lt;br /&gt;
&lt;br /&gt;
Integration Port = Leave blank and the software uses 80 to verify its license with the PBX.&amp;amp;nbsp; If the softphone is remote and an external port other than 80 is forwarded to the PBX, enter that as the Integration Port (eg, 8080 external forwarded to 80 internal)&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
All other information can be left at Default.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Communicator-SIP Acc.jpg|File:Communicator-SIP Acc.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Be Sure to Click Enabled when you are returned to the account screen on the account you just created so it is made the live account&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone8.jpg|File:SoftPhone8.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Click close and you should be returned to the User interface and your account should register to the PBX.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone9.jpg|File:SoftPhone9.jpg]][[File:SoftPhone10.jpg|File:SoftPhone10.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Your Softphone is now ready to be used.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Import Extensions from the PBX to Contacts.'''&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Import all Extension in the PBX into the Contact list select User List - Manage Users - Ipitomy and then Refresh&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone11.jpg|File:SoftPhone11.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Select the PBX from the Host list&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone14.jpg|File:SoftPhone14.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Click OK. The Softphone will now pull down an Extension list from the PBX which you configured.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone13.jpg|File:SoftPhone13.jpg]]&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPitomy_Communicator&amp;diff=5123</id>
		<title>IPitomy Communicator</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPitomy_Communicator&amp;diff=5123"/>
		<updated>2024-12-10T15:29:34Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Note: This Guide will assist the user in Installing and Using the IPitomy Communicator Softphone.&lt;br /&gt;
&lt;br /&gt;
We have two different soft phone installs. The “demo” version does not include the user list which shows all of the presence of all of the users on the system. This is intended for trial versions or for users who are on a shared instance of IPitomy in the Cloud. For obvious reasons, on a shared instance, it would not be good to display the presence of all the other users!&lt;br /&gt;
&lt;br /&gt;
The full commercial version is called IPitomy Communicator.&lt;br /&gt;
&lt;br /&gt;
This requires an available IPitomy Extension License as well as an IPitomy Soft Phone License. This does not have auto configuration, but is super simple to set up. When installing it, select IP620 as the device type for the extension so it will use an IPitomy license instead of an open license.&lt;br /&gt;
&lt;br /&gt;
This also requires the generation of a single API Key. If your PBX already has one generated for the IPitomy Communicator, you don't need to do anything else. If not, simply navigate to Applications=&amp;gt;API Key, name it Softphone [one word, capital S], and click Create.&amp;amp;nbsp; Be sure to enable it by checking the box labeled &amp;quot;Key Enable&amp;quot;, then save and apply changes.&lt;br /&gt;
&lt;br /&gt;
Links for SoftPhone:&lt;br /&gt;
[[File:Screenshot for communicator.png|alt=Screenshot for communicator|thumb|518x518px|Screenshot for communicator]]&lt;br /&gt;
[[File:Screenshot for communicator 2.png|thumb|519x519px|Screenshot for communicator 2]]&lt;br /&gt;
Demo version: [http://relay.ipitomy.com/softcall_demo/publish.htm http://relay.ipitomy.com/softcall_demo/publish.htm]&lt;br /&gt;
&lt;br /&gt;
Full version – IPitomy Communicator: [https://www.dropbox.com/s/6uhj12mwh8k7diu/CommunicatorSetup.zip?dl=0 &amp;lt;nowiki&amp;gt;[Here]&amp;lt;/nowiki&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''''&amp;lt;big&amp;gt;&amp;lt;u&amp;gt;IMPORTANT&amp;lt;/u&amp;gt;&amp;lt;/big&amp;gt;'''''&lt;br /&gt;
&lt;br /&gt;
To avoid an error, you must also install ''Visual C++ Redistributable for Visual Studio 2012 Update 4''.&lt;br /&gt;
&lt;br /&gt;
The file is here:  '''''&amp;lt;big&amp;gt;https://www.microsoft.com/en-us/download/details.aspx?id=30679&amp;lt;/big&amp;gt;'''''&lt;br /&gt;
&lt;br /&gt;
Scroll down just a little bit, click the download button, then select the x86 version (yes, even if you're running 64 bit Windows).&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== '''Ipitomy Softphone'''&amp;lt;br/&amp;gt; ==&lt;br /&gt;
&lt;br /&gt;
'''Installation'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Installation Video'''&lt;br /&gt;
&lt;br /&gt;
[https://www.youtube.com/watch?v=p5gwURMp4eo &amp;lt;b&amp;gt;https://www.youtube.com/watch?v=p5gwURMp4eo&amp;lt;/b&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Park Button Video'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[https://www.youtube.com/watch?v=dPCVXiR6kf8 &amp;lt;b&amp;gt;https://www.youtube.com/watch?v=dPCVXiR6kf8&amp;lt;/b&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Install the Softphone, Follow the URL above and click the Installer link. Installation will begin at that time. You should have adminstrative rights to the PC you are installing to prior to installation. Accept the default location for install unless you have some other directory you wish to install to.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Setup'''&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Once isntalled open the program by clicking the program Icon. This should now be located on your Desktop as well as in the Program directory. The Program will open bringing up the user interface screen.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone3.jpg|File:SoftPhone3.jpg]][[File:SoftPhone4.jpg|File:SoftPhone4.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Add an account Press the Settings Icon which is the little Gear Icon next to the Ipitomy Logo Bar. Select Sip Accounts and Add an Account&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone5.jpg|File:SoftPhone5.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Select the Add Button and Enter the Following:&lt;br /&gt;
&lt;br /&gt;
Display Name = Name of user&lt;br /&gt;
&lt;br /&gt;
User Name = Extension Number&lt;br /&gt;
&lt;br /&gt;
User Password = Voicemail Password&lt;br /&gt;
&lt;br /&gt;
SIP Password = SIP Password for Extension&lt;br /&gt;
&lt;br /&gt;
SIP Server = IP of PBX (Local IP is OK for a Local Extension only, however for a remote phone this would need to the Public IP of the Network the PBX resides upon)&lt;br /&gt;
&lt;br /&gt;
Integration Port = Leave blank and the software uses 80 to verify its license with the PBX.&amp;amp;nbsp; If the softphone is remote and an external port other than 80 is forwarded to the PBX, enter that as the Integration Port (eg, 8080 external forwarded to 80 internal)&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
All other information can be left at Default.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Communicator-SIP Acc.jpg|File:Communicator-SIP Acc.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Be Sure to Click Enabled when you are returned to the account screen on the account you just created so it is made the live account&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone8.jpg|File:SoftPhone8.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Click close and you should be returned to the User interface and your account should register to the PBX.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone9.jpg|File:SoftPhone9.jpg]][[File:SoftPhone10.jpg|File:SoftPhone10.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Your Softphone is now ready to be used.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Import Extensions from the PBX to Contacts.'''&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Import all Extension in the PBX into the Contact list select User List - Manage Users - Ipitomy and then Refresh&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone11.jpg|File:SoftPhone11.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Select the PBX from the Host list&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone14.jpg|File:SoftPhone14.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Click OK. The Softphone will now pull down an Extension list from the PBX which you configured.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone13.jpg|File:SoftPhone13.jpg]]&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPitomy_Communicator&amp;diff=5122</id>
		<title>IPitomy Communicator</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPitomy_Communicator&amp;diff=5122"/>
		<updated>2024-12-10T15:28:56Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: added a link and instructions for Visual C++ Redistributable for Visual Studio 2012 Update 4&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Note: This Guide will assist the user in Installing and Using the IPitomy Communicator Softphone.&lt;br /&gt;
&lt;br /&gt;
We have two different soft phone installs. The “demo” version does not include the user list which shows all of the presence of all of the users on the system. This is intended for trial versions or for users who are on a shared instance of IPitomy in the Cloud. For obvious reasons, on a shared instance, it would not be good to display the presence of all the other users!&lt;br /&gt;
&lt;br /&gt;
The full commercial version is called IPitomy Communicator.&lt;br /&gt;
&lt;br /&gt;
This requires an available IPitomy Extension License as well as an IPitomy Soft Phone License. This does not have auto configuration, but is super simple to set up. When installing it, select IP620 as the device type for the extension so it will use an IPitomy license instead of an open license.&lt;br /&gt;
&lt;br /&gt;
This also requires the generation of a single API Key. If your PBX already has one generated for the IPitomy Communicator, you don't need to do anything else. If not, simply navigate to Applications=&amp;gt;API Key, name it Softphone [one word, capital S], and click Create.&amp;amp;nbsp; Be sure to enable it by checking the box labeled &amp;quot;Key Enable&amp;quot;, then save and apply changes.&lt;br /&gt;
&lt;br /&gt;
Links for SoftPhone:&lt;br /&gt;
[[File:Screenshot for communicator.png|alt=Screenshot for communicator|thumb|518x518px|Screenshot for communicator]]&lt;br /&gt;
[[File:Screenshot for communicator 2.png|thumb|519x519px|Screenshot for communicator 2]]&lt;br /&gt;
Demo version: [http://relay.ipitomy.com/softcall_demo/publish.htm http://relay.ipitomy.com/softcall_demo/publish.htm]&lt;br /&gt;
&lt;br /&gt;
Full version – IPitomy Communicator: [https://www.dropbox.com/s/6uhj12mwh8k7diu/CommunicatorSetup.zip?dl=0 &amp;lt;nowiki&amp;gt;[Here]&amp;lt;/nowiki&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
'''''&amp;lt;big&amp;gt;&amp;lt;u&amp;gt;IMPORTANT&amp;lt;/u&amp;gt;&amp;lt;/big&amp;gt;'''''&lt;br /&gt;
&lt;br /&gt;
To avoid an error, you must also install ''Visual C++ Redistributable for Visual Studio 2012 Update 4''.&lt;br /&gt;
&lt;br /&gt;
The file is here:  '''''&amp;lt;big&amp;gt;https://www.microsoft.com/en-us/download/details.aspx?id=30679&amp;lt;/big&amp;gt;'''''&lt;br /&gt;
&lt;br /&gt;
Scroll down just a little bit, click the download button, then select the x86 version (yes, even if you're running 64 bit Windows).&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== '''Ipitomy Softphone'''&amp;lt;br/&amp;gt; ==&lt;br /&gt;
&lt;br /&gt;
'''Installation'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Installation Video'''&lt;br /&gt;
&lt;br /&gt;
[https://www.youtube.com/watch?v=p5gwURMp4eo &amp;lt;b&amp;gt;https://www.youtube.com/watch?v=p5gwURMp4eo&amp;lt;/b&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Park Button Video'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[https://www.youtube.com/watch?v=dPCVXiR6kf8 &amp;lt;b&amp;gt;https://www.youtube.com/watch?v=dPCVXiR6kf8&amp;lt;/b&amp;gt;]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Install the Softphone, Follow the URL above and click the Installer link. Installation will begin at that time. You should have adminstrative rights to the PC you are installing to prior to installation. Accept the default location for install unless you have some other directory you wish to install to.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Setup'''&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Once isntalled open the program by clicking the program Icon. This should now be located on your Desktop as well as in the Program directory. The Program will open bringing up the user interface screen.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone3.jpg|File:SoftPhone3.jpg]][[File:SoftPhone4.jpg|File:SoftPhone4.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Add an account Press the Settings Icon which is the little Gear Icon next to the Ipitomy Logo Bar. Select Sip Accounts and Add an Account&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone5.jpg|File:SoftPhone5.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Select the Add Button and Enter the Following:&lt;br /&gt;
&lt;br /&gt;
Display Name = Name of user&lt;br /&gt;
&lt;br /&gt;
User Name = Extension Number&lt;br /&gt;
&lt;br /&gt;
User Password = Voicemail Password&lt;br /&gt;
&lt;br /&gt;
SIP Password = SIP Password for Extension&lt;br /&gt;
&lt;br /&gt;
SIP Server = IP of PBX (Local IP is OK for a Local Extension only, however for a remote phone this would need to the Public IP of the Network the PBX resides upon)&lt;br /&gt;
&lt;br /&gt;
Integration Port = Leave blank and the software uses 80 to verify its license with the PBX.&amp;amp;nbsp; If the softphone is remote and an external port other than 80 is forwarded to the PBX, enter that as the Integration Port (eg, 8080 external forwarded to 80 internal)&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
All other information can be left at Default.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Communicator-SIP Acc.jpg|File:Communicator-SIP Acc.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Be Sure to Click Enabled when you are returned to the account screen on the account you just created so it is made the live account&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone8.jpg|File:SoftPhone8.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Click close and you should be returned to the User interface and your account should register to the PBX.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone9.jpg|File:SoftPhone9.jpg]][[File:SoftPhone10.jpg|File:SoftPhone10.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Your Softphone is now ready to be used.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Import Extensions from the PBX to Contacts.'''&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
To Import all Extension in the PBX into the Contact list select User List - Manage Users - Ipitomy and then Refresh&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone11.jpg|File:SoftPhone11.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Select the PBX from the Host list&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone14.jpg|File:SoftPhone14.jpg]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Click OK. The Softphone will now pull down an Extension list from the PBX which you configured.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:SoftPhone13.jpg|File:SoftPhone13.jpg]]&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=File:Screenshot_for_communicator_2.png&amp;diff=5121</id>
		<title>File:Screenshot for communicator 2.png</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=File:Screenshot_for_communicator_2.png&amp;diff=5121"/>
		<updated>2024-12-10T15:22:50Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Screenshot for communicator 2&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=File:Screenshot_for_communicator.png&amp;diff=5120</id>
		<title>File:Screenshot for communicator.png</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=File:Screenshot_for_communicator.png&amp;diff=5120"/>
		<updated>2024-12-10T15:15:59Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Screenshot for communicator&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=Call_Recording&amp;diff=5119</id>
		<title>Call Recording</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=Call_Recording&amp;diff=5119"/>
		<updated>2024-11-27T17:28:16Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;IMPORTANT: In many jurisdictions, it is a requirement to announce the call is being recorded.&amp;amp;nbsp; To that end, you can insert this prompt into either a menu that fails over to the recorded ring group, or if you have ACD, as an intro announcement on the acd group being recorded itself:&amp;amp;nbsp;&amp;amp;nbsp;[https://www.dropbox.com/s/nplru2ea5iqtxgn/this-call-may-be-monitored-or-recorded.wav?dl=0 Call May Be Recorded audio file]&lt;br /&gt;
&lt;br /&gt;
This feature allows recording of inbound and outbound calls. Extensions can be put into groups which makes it easier to keep track of calls. There can be jobs that only record inbound calls, outbound calls, or both.&lt;br /&gt;
&lt;br /&gt;
==Recording Methods==&lt;br /&gt;
&lt;br /&gt;
=== Manual Recordings: &amp;amp;nbsp; ===&lt;br /&gt;
*Initiated by agent, or Extension User while on call by dialing *#*# or pressing a programmed key (a BLF set to speed dial) on the phone.&lt;br /&gt;
**These recordings are either in Agent Recordings (if they are recordings by an agent taking a queue call). &amp;amp;nbsp;Or they are available in the work folder in the recording extension's mailbox.&amp;amp;nbsp;&lt;br /&gt;
**You will need to ensure that the Call Recording is enabled in the proper places for this to work:&lt;br /&gt;
***Trunks&lt;br /&gt;
***Ring Groups&lt;br /&gt;
***Extension initiating the recording&lt;br /&gt;
***Extension being recorded&lt;br /&gt;
***Extension Advanced settings must have &amp;quot;Is Operator&amp;quot; checked.&lt;br /&gt;
=== Initiated by QManager ===&lt;br /&gt;
**These recordings are available through View Recordings in QManager&lt;br /&gt;
=== Third Party Recording ===&lt;br /&gt;
*IPitomy has implemented integration with Trivium to support their call recording application.&lt;br /&gt;
=== IPitomy Call Recording Application ===&lt;br /&gt;
*Licensed IPitomy Feature that records all calls on designated destinations.&lt;br /&gt;
**NOTE: This cannot record Conference Rooms&lt;br /&gt;
&lt;br /&gt;
NOTE: When using the IPitomy Call Recording feature, it is advised to configure an FTP to push those recordings to so they can be keep for a long time, and to set the recording job to purge after an FTP push so that the file size does not get too large. [[File:Callrecording.png|center|Callrecording.png]]&lt;br /&gt;
&lt;br /&gt;
== Create New Recording Job ==&lt;br /&gt;
&lt;br /&gt;
#Navigate to the Call Recording Page under the Applications tab.&lt;br /&gt;
#Click Add Recording&lt;br /&gt;
#Enter the name of the recording group.&lt;br /&gt;
#Add Ring Groups to the recording job that are applicable to the group that you want to be recorded.&lt;br /&gt;
#Add Available Routes that you would like to be recorded.&lt;br /&gt;
#Click on the [[File:Savechanges.png]] button.&lt;br /&gt;
&amp;lt;p style=&amp;quot;text-align: center&amp;quot;&amp;gt;[[File:RecordingJob.jpg|File:RecordingJob.jpg]]&amp;lt;/p&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{| class=&amp;quot;wikitable&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
! Sections/Fields&lt;br /&gt;
! Description&lt;br /&gt;
|-&lt;br /&gt;
| colspan=&amp;quot;2&amp;quot; | '''Recording Settings'''&lt;br /&gt;
|-&lt;br /&gt;
| Name&lt;br /&gt;
| Enter the name of the Recording Job&lt;br /&gt;
|-&lt;br /&gt;
| Enable&lt;br /&gt;
| Option to enable or disable call recording&lt;br /&gt;
|-&lt;br /&gt;
| Maximum number of recordings&lt;br /&gt;
| Enter the maximum amount of calls you would like to have recorded. If the maximum number has been reached, then the old recordings will be deleted to make room for new ones.&lt;br /&gt;
|-&lt;br /&gt;
| Maximum age of recordings&lt;br /&gt;
| Enter the maximum age of recordings. This will delete recordings after a certain period of time. This will clear out old recordings automatically to make room for new recordings.&lt;br /&gt;
|-&lt;br /&gt;
| Available Ring Groups&lt;br /&gt;
| These are the groups that can be selected to include in the recording job&lt;br /&gt;
|-&lt;br /&gt;
| Selected&lt;br /&gt;
| These are the groups that have been added to be included in the recording job. These are the extensions that you will find recordings for when checking this recording job&lt;br /&gt;
|-&lt;br /&gt;
| Available Routes&lt;br /&gt;
| Routes that can be selected to be included in the recording job. When one of these routes is dialed by any extensions on the system, the call will be recorded.&lt;br /&gt;
|-&lt;br /&gt;
| Selected&lt;br /&gt;
| These are the routes that have been selected to be recorded when dialed by the selected ring groups.&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
=== Viewing Recording Jobs ===&lt;br /&gt;
&lt;br /&gt;
[[File:Managerecordingfiles.png|center|Managerecordingfiles.png]]&lt;br /&gt;
&lt;br /&gt;
To View the recording jobs and manage or listen to the recorded calls:&lt;br /&gt;
&lt;br /&gt;
#Navigate to the Call Recording page under the Applications tab.&lt;br /&gt;
#Click the [[File:Viewjobs.png]] icon to view the recording files for the group you wish to view.&lt;br /&gt;
#To download/listen to the recording files, click the [[File:Downloadbutton.jpg]] button.&lt;br /&gt;
&lt;br /&gt;
=== Archived Recordings ===&lt;br /&gt;
&lt;br /&gt;
After a day, the PBX will archive the recordings from the previous day. These are put into a .zip folder and can be viewed on the PBX or downloaded. If you download them, there will be an index.html file contained within the .zip that gives information about each recording (Recording Target, Filename, Size, Date, Source &amp;amp; Destination) so you can find the recording you are looking for. You can also then click the link and it will play the file you want.&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPitomy_Contact_Dialer&amp;diff=5118</id>
		<title>IPitomy Contact Dialer</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPitomy_Contact_Dialer&amp;diff=5118"/>
		<updated>2024-11-13T16:00:22Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;__TOC__&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Ipitomy Contact Dialer'''&lt;br /&gt;
&lt;br /&gt;
To set up the IPitomy Dialer you must first be on the latest (5.0.7-X) PBX software.&amp;lt;br/&amp;gt;&amp;lt;br/&amp;gt;Dialer requires .NET Framework 4.0 or greater. Windows 7 or newer Microsoft OS. You also need to be licensed for the applications use. This license may be obtained from Ipitomy Sales if you do not already have it. Until this license is applied to the PBX you will not have the ability to create the Feature Key.&lt;br /&gt;
&lt;br /&gt;
'''Note'''&lt;br /&gt;
If you are having issues launching the Contact Dialer and have verified your information is correct, ensure the Time on your PC is accurate.  If the time is off from that of the PBX it will give a 408 Timeout error.  You also may need to ensure that the Daylight Savings Time box is checked in Windows or you may also get the 408 Timeout error.&lt;br /&gt;
&lt;br /&gt;
The application can be downloaded from:&lt;br /&gt;
&lt;br /&gt;
[http://relay.ipitomy.com/contacts/Contacts.application Click here to download the installation file]&lt;br /&gt;
&lt;br /&gt;
''&amp;lt;small&amp;gt;If clicking the link does not initiate the download, try a different browser. You can also right click, select &amp;quot;copy link address&amp;quot; then paste the link address into another browser tab.&amp;lt;/small&amp;gt;''&lt;br /&gt;
&lt;br /&gt;
Once you have upgraded you will have under the Applications area of the PBX GUI an API Page.&lt;br /&gt;
&lt;br /&gt;
=== Installation Video ===&lt;br /&gt;
&lt;br /&gt;
https://www.youtube.com/watch?v=6ginRqzHcNc&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;Use the tool to create a KEY named &amp;quot;Contacts&amp;quot; (case sensitive)&lt;br /&gt;
&lt;br /&gt;
[[File:Ipitomy Dialer1.jpg|File:Ipitomy Dialer1.jpg]]&lt;br /&gt;
&lt;br /&gt;
Once finished your Key will look like this.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:IpDialer.jpg|File:IpDialer.jpg]]&lt;br /&gt;
&lt;br /&gt;
At this point you can use the guide on the following link to configure and use the dialer.&lt;br /&gt;
&lt;br /&gt;
[http://wiki.ipitomy.com/wiki/File:IPitomy_Dialer_App.pdf http://wiki.ipitomy.com/wiki/File:IPitomy_Dialer_App.pdf]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;The installation will begin. When finished it will bring you to the configuration page.&lt;br /&gt;
&lt;br /&gt;
[[File:Contact 1.jpg|File:Contact 1.jpg]]&lt;br /&gt;
&lt;br /&gt;
Enter the Extension number where indicated. The Password used is the Extension Voicemail Password. And then populate the IP of the PBX in the appropriate field. No changeto the port number is required.&lt;br /&gt;
&lt;br /&gt;
This will bring up the main page.&lt;br /&gt;
&lt;br /&gt;
[[File:Contact 2.png|File:Contact 2.png]]&lt;br /&gt;
&lt;br /&gt;
System Speed dials are present. To dial these simply click on them. The device you configured will ring. And once you pick up the call will be connecting. You can import contacts from a CSV file or from a local Outlook installation. You can also create manually the entries you desire. Enter the Extension number.&lt;br /&gt;
&lt;br /&gt;
'''&amp;amp;nbsp;IPitomy Dialer'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;&amp;lt;br/&amp;gt;'''Overview'''&amp;lt;br/&amp;gt;Many users of business phone systems call a wide variety of phone numbers. They can be from a printed sheet, a contact program or from Microsoft Outlook. When there are many calls to make and each number has to be dialed into a phone, the process becomes tedious. To ease the process of dialing each number, IPitomy has created the IPitomy Dialer. The&amp;lt;br/&amp;gt;dialer can be used in several ways:&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;1) Automatically import contacts from Microsoft Outlook. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;2) Import Contacts from a .csv (comma separated values) file. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;3) Dialing System speed dial numbers. The PBX will automatically import any system speed dial numbers. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;4) Click to dial from Mozilla Firefox. using the Telify add-on, phone numbers in web pages can be clicked to dial.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;5) Clip to Dial - Copy - by selecting a phone number in any program running on your desktop, Select the phone number by highlighting it with a mouse. Click Control C then Control D and the number is dialed. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
=== Program Setup ===&lt;br /&gt;
&lt;br /&gt;
You first need to add your extension and Pin number to the Dialer in order for the IPitomyPBX to dial from your extension. Click Settings in the upper left corner Enter your extension number in the Extension field Enter the PIN number you received for your Voice mail Box Add the IPitomy server IPaddress. You will get this from your system administrator. Click Connect Now the Dialer is setup to connect with the PBX and dial out numbers from your desktop.Before you can begin dialing, it is necessary to populate the contact dialer with contacts so you can dial them with just a click. To import Contacts from Microsoft Outlook, you must first be using Outlook 2007 or newer. Click on Advanced under the Managed Contacts tab. After clicking the Advanced tab, you will see the option to import Outlook Contacts. Click the Import Outlook Contacts tab, once this is clicked, the Dialer will import all of your contacts. It may take up to 10 minutes, depending on how many contacts you have&lt;br /&gt;
&lt;br /&gt;
=== Importing Contacts ===&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;From Outlook, Click Manage Contacts. Click Import Click Import Outlook Contacts. The Dialer will import all of your Outlook Contacts. Once imported they will appear in a list. just click on the item in the list you wish to dial.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Importing Contacts From a .csv File'''&amp;lt;br/&amp;gt;Click Manage Contacts Click Import Click Import from CSV file Older versions of outlook and Combinations of contacts from a .csv file contacts and outlook contacts. In order to import contacts from older versions of Outlook, it may be necessary to first export your Outlook Contacts to a .csv file. Then import the file using the .csv import function.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Using The Import From .csv Function'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;To import contacts from a .csv file, first, make sure your .CSV file has a header line. '''''The Contact dialer will ignore the first row of the .csv'''''. It is necessary to map the fields in the contact from the .csv file to match up the fields in the IPitomy Dialer contact fields. To do this, you click on Import Contacts from CSV Then browse to the file you want to import on your PC Select the file and click Open The file will then begin the import process. It is now necessary to map the fields from your .csv file to the fields in the IPitomy Dialer. Place your mouse over the column field in the left column that best matches the field on the Right side and drag it over to the right column. Once you have matched up the fields to want to map, then click Import. That’s all there is to it. You can update the list any time by re-importing it or manually adding the contact entries. Once you have imported the contacts, your list will be populated with the contact information and the dialer will dial the numbers when you select a contact When you double click a contact, the number is dialed. Your phone will go off hook (if you are using a phone that is capable) and your number is dialed.&lt;br /&gt;
&lt;br /&gt;
=== Integration with Telify ===&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{{:Telify}}&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPitomy_Contact_Dialer&amp;diff=5117</id>
		<title>IPitomy Contact Dialer</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPitomy_Contact_Dialer&amp;diff=5117"/>
		<updated>2024-11-13T15:59:05Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;__TOC__&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Ipitomy Contact Dialer'''&lt;br /&gt;
&lt;br /&gt;
To set up the IPitomy Dialer you must first be on the latest (5.0.7-X) PBX software.&amp;lt;br/&amp;gt;&amp;lt;br/&amp;gt;Dialer requires .NET Framework 4.0 or greater. Windows 7 or newer Microsoft OS. You also need to be licensed for the applications use. This license may be obtained from Ipitomy Sales if you do not already have it. Until this license is applied to the PBX you will not have the ability to create the Feature Key.&lt;br /&gt;
&lt;br /&gt;
'''Note'''&lt;br /&gt;
If you are having issues launching the Contact Dialer and have verified your information is correct, ensure the Time on your PC is accurate.  If the time is off from that of the PBX it will give a 408 Timeout error.  You also may need to ensure that the Daylight Savings Time box is checked in Windows or you may also get the 408 Timeout error.&lt;br /&gt;
&lt;br /&gt;
The application can be downloaded from:&lt;br /&gt;
&lt;br /&gt;
[http://relay.ipitomy.com/contacts/Contacts.application Click here to download the installation file]&lt;br /&gt;
&lt;br /&gt;
''&amp;lt;small&amp;gt;If clicking the link does not initiate the download, try a different browser. You can also right click, select &amp;quot;copy link address&amp;quot; then paste the link address into another browser tab.&amp;lt;/small&amp;gt;''&lt;br /&gt;
&lt;br /&gt;
Once you have upgraded you will have under the Applications area of the PBX GUI an API Page.&lt;br /&gt;
&lt;br /&gt;
=== Installation Video ===&lt;br /&gt;
&lt;br /&gt;
https://www.youtube.com/watch?v=6ginRqzHcNc&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;Use the tool to create a KEY named &amp;quot;Contacts&amp;quot; (case sensitive)&lt;br /&gt;
&lt;br /&gt;
[[File:Ipitomy Dialer1.jpg|File:Ipitomy Dialer1.jpg]]&lt;br /&gt;
&lt;br /&gt;
Once finished your Key will look like this.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:IpDialer.jpg|File:IpDialer.jpg]]&lt;br /&gt;
&lt;br /&gt;
At this point you can use the guide on the following link to configure and use the dialer.&lt;br /&gt;
&lt;br /&gt;
[http://wiki.ipitomy.com/wiki/File:IPitomy_Dialer_App.pdf http://wiki.ipitomy.com/wiki/File:IPitomy_Dialer_App.pdf]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;The installation will begin. When finished it will bring you to the configuration page.&lt;br /&gt;
&lt;br /&gt;
[[File:Contact 1.jpg|File:Contact 1.jpg]]&lt;br /&gt;
&lt;br /&gt;
Enter the Extension number where indicated. The Password used is the Extension Voicemail Password. And then populate the IP of the PBX in the appropriate field. No changeto the port number is required.&lt;br /&gt;
&lt;br /&gt;
This will bring up the main page.&lt;br /&gt;
&lt;br /&gt;
[[File:Contact 2.png|File:Contact 2.png]]&lt;br /&gt;
&lt;br /&gt;
System Speed dials are present. To dial these simply click on them. The device you configured will ring. And once you pick up the call will be connecting. You can import contacts from a CSV file or from a local Outlook installation. You can also create manually the entries you desire. Enter the Extension number.&lt;br /&gt;
&lt;br /&gt;
'''&amp;amp;nbsp;IPitomy Dialer'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;&amp;lt;br/&amp;gt;'''Overview'''&amp;lt;br/&amp;gt;Many users of business phone systems call a wide variety of phone numbers. They can be from a printed sheet, a contact program or from Microsoft Outlook. When there are many calls to make and each number has to be dialed into a phone, the process becomes tedious. To ease the process of dialing each number, IPitomy has created the IPitomy Dialer. The&amp;lt;br/&amp;gt;dialer can be used in several ways:&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;1) Automatically import contacts from Microsoft Outlook. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;2) Import Contacts from a .csv (comma separated values) file. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;3) Dialing System speed dial numbers. The PBX will automatically import any system speed dial numbers. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;4) Click to dial from Mozilla Firefox. using the Telify add-on, phone numbers in web pages can be clicked to dial.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;5) Clip to Dial - Copy - by selecting a phone number in any program running on your desktop, Select the phone number by highlighting it with a mouse. Click Control C then Control D and the number is dialed. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
=== Program Setup ===&lt;br /&gt;
&lt;br /&gt;
You first need to add your extension and Pin number to the Dialer in order for the IPitomyPBX to dial from your extension. Click Settings in the upper left corner Enter your extension number in the Extension field Enter the PIN number you received for your Voice mail Box Add the IPitomy server IPaddress. You will get this from your system administrator. Click Connect Now the Dialer is setup to connect with the PBX and dial out numbers from your desktop.Before you can begin dialing, it is necessary to populate the contact dialer with contacts so you can dial them with just a click. To import Contacts from Microsoft Outlook, you must first be using Outlook 2007 or newer. Click on Advanced under the Managed Contacts tab. After clicking the Advanced tab, you will see the option to import Outlook Contacts. Click the Import Outlook Contacts tab, once this is clicked, the Dialer will import all of your contacts. It may take up to 10 minutes, depending on how many contacts you have&lt;br /&gt;
&lt;br /&gt;
=== Importing Contacts ===&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;From Outlook Click Manage Contacts Click Import Click Import Outlook ContactsThe Dialer will import all of your Outlook Contacts. Once imported they will appear ina list. just click on the item in the list you wish to dial.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Importing Contacts From a .csv File'''&amp;lt;br/&amp;gt;Click Manage Contacts Click Import Click Import from CSV file Older versions of outlook and Combinations of contacts from a .csv file contacts and outlook contacts. In order to import contacts from older versions of Outlook, it may be necessary to first export your Outlook Contacts to a .csv file. Then import the file using the .csv import function.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Using The Import From .csv Function'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;To import contacts from a .csv file, first, make sure your .CSV file has a header line. '''''The Contact dialer will ignore the first row of the .csv'''''. It is necessary to map the fields in the contact from the .csv file to match up the fields in the IPitomy Dialer contact fields. To do this, you click on Import Contacts from CSV Then browse to the file you want to import on your PC Select the file and click Open The file will then begin the import process. It is now necessary to map the fields from your .csv file to the fields in the IPitomy Dialer. Place your mouse over the column field in the left column that best matches the field on the Right side and drag it over to the right column. Once you have matched up the fields to want to map, then click Import. That’s all there is to it. You can update the list any time by re-importing it or manually adding the contact entries. Once you have imported the contacts, your list will be populated with the contact information and the dialer will dial the numbers when you select a contact When you double click a contact, the number is dialed. Your phone will go off hook (if you are using a phone that is capable) and your number is dialed.&lt;br /&gt;
&lt;br /&gt;
=== Integration with Telify ===&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{{:Telify}}&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPitomy_Contact_Dialer&amp;diff=5116</id>
		<title>IPitomy Contact Dialer</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPitomy_Contact_Dialer&amp;diff=5116"/>
		<updated>2024-11-13T15:57:41Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;__TOC__&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Ipitomy Contact Dialer'''&lt;br /&gt;
&lt;br /&gt;
To set up the IPitomy Dialer you must first be on the latest (5.0.7-X) PBX software.&amp;lt;br/&amp;gt;&amp;lt;br/&amp;gt;Dialer requires .NET Framework 4.0 or greater. Windows 7 or newer Microsoft OS. You also need to be licensed for the applications use. This license may be obtained from Ipitomy Sales if you do not already have it. Until this license is applied to the PBX you will not have the ability to create the Feature Key.&lt;br /&gt;
&lt;br /&gt;
'''Note'''&lt;br /&gt;
If you are having issues launching the Contact Dialer and have verified your information is correct, ensure the Time on your PC is accurate.  If the time is off from that of the PBX it will give a 408 Timeout error.  You also may need to ensure that the Daylight Savings Time box is checked in Windows or you may also get the 408 Timeout error.&lt;br /&gt;
&lt;br /&gt;
The application can be downloaded from:&lt;br /&gt;
&lt;br /&gt;
[http://relay.ipitomy.com/contacts/Contacts.application Click here to download the installation file]&lt;br /&gt;
&lt;br /&gt;
''&amp;lt;small&amp;gt;If clicking the link does not initiate the download, try a different browser. You can also right click, select &amp;quot;copy link address&amp;quot; then paste the link address into another browser tab.&amp;lt;/small&amp;gt;''&lt;br /&gt;
&lt;br /&gt;
Once you have upgraded you will have under the Applications area of the PBX GUI an API Page.&lt;br /&gt;
&lt;br /&gt;
=== Installation Video ===&lt;br /&gt;
&lt;br /&gt;
https://www.youtube.com/watch?v=6ginRqzHcNc&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;Use the tool to create a KEY named &amp;quot;Contacts&amp;quot; (case sensitive)&lt;br /&gt;
&lt;br /&gt;
[[File:Ipitomy Dialer1.jpg|File:Ipitomy Dialer1.jpg]]&lt;br /&gt;
&lt;br /&gt;
Once finished your Key will look like this.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:IpDialer.jpg|File:IpDialer.jpg]]&lt;br /&gt;
&lt;br /&gt;
At this point you can use the guide on the following link to configure and use the dialer.&lt;br /&gt;
&lt;br /&gt;
[http://wiki.ipitomy.com/wiki/File:IPitomy_Dialer_App.pdf http://wiki.ipitomy.com/wiki/File:IPitomy_Dialer_App.pdf]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;The installation will begin. When finished it will bring you to the configuration page.&lt;br /&gt;
&lt;br /&gt;
[[File:Contact 1.jpg|File:Contact 1.jpg]]&lt;br /&gt;
&lt;br /&gt;
Enter the Extension number where indicated. The Password used is the Extension Voicemail Password. And then populate the IP of the PBX in the appropriate field. No changeto the port number is required.&lt;br /&gt;
&lt;br /&gt;
This will bring up the main page.&lt;br /&gt;
&lt;br /&gt;
[[File:Contact 2.png|File:Contact 2.png]]&lt;br /&gt;
&lt;br /&gt;
System Speed dials are present. To dial these simply click on them. The device you configured will ring. And once you pick up the call will be connecting. You can import contacts from a CSV file or from a local Outlook installation. You can also create manually the entries you desire. Enter the Extension number.&lt;br /&gt;
&lt;br /&gt;
'''&amp;amp;nbsp;IPitomy Dialer'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;&amp;lt;br/&amp;gt;'''Overview'''&amp;lt;br/&amp;gt;Many users of business phone systems call a wide variety of phone numbers. They can be from a printed sheet, a contact program or from Microsoft Outlook. When there are many calls to make and each number has to be dialed into a phone, the process becomes tedious. To ease the process of dialing each number, IPitomy has created the IPitomy Dialer. The&amp;lt;br/&amp;gt;dialer can be used in several ways:&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;1) Automatically import contacts from Microsoft Outlook. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;2) Import Contacts from a .csv (comma separated values) file. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;3) Dialing System speed dial numbers. The PBX will automatically import any system speed dial numbers. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;4) Click to dial from Mozilla Firefox. using the Telify add-on, phone numbers in web pages can be clicked to dial.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;5) Clip to Dial - Copy - by selecting a phone number in any program running on your desktop, Select the phone number by highlighting it with a mouse. Click Control C then Control D and the number is dialed. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
=== Program Setup ===&lt;br /&gt;
&lt;br /&gt;
You first need to add your extension and Pin number to the Dialer in order for the IPitomyPBX to dial from your extension. Click Settings in the upper left corner Enter your extension number in the Extension field Enter the PIN number you received for your Voice mail Box Add the IPitomy server IPaddress. You will get this from your system administrator. Click Connect Now the Dialer is setup to connect with the PBX and dial out numbers from your desktop.Before you can begin dialing, it is necessary to populate the contact dialer with contacts so you can dial them with just a click. To import Contacts from Microsoft Outlook, you must first be using Outlook 2007 or newer. Click on Advanced under the Managed Contacts tab. After clicking the Advanced tab, you will see the option to import Outlook Contacts. Click the Import Outlook Contacts tab, once this is clicked, the Dialer will import all of your contacts. It may take up to 10 minutes, depending on how many contacts you have&lt;br /&gt;
&lt;br /&gt;
=== Importing Contacts ===&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;From Outlook Click Manage Contacts Click Import Click Import Outlook ContactsThe Dialer will import all of your Outlook Contacts. Once imported they will appear ina list. just click on the item in the list you wish to dial.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Importing Contacts From a .csv File'''&amp;lt;br/&amp;gt;Click Manage Contacts Click Import Click Import from CSV file Older versions of outlook and Combinations of contacts from a .csv file contacts and outlook contacts. In order to import contacts from older versions of Outlook, it may be necessary to first export your Outlook Contacts to a .csv file. Then import the file using the .csv import function.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Using The Import From .csv Function'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;To import contacts from a .csv file, first, make sure your .CSV file has a header line. '''''The Contact dialer will ignore the first row of the .CSV'''''. It is necessary to map the fields in the contact from the .csv file to match up the fields in the IPitomy Dialer contact fields. To do this, you click on Import Contacts from CSV Then browse to the file you want to import on your PC Select the file and click Open The file will then begin the import process. It is now necessary to map the fields from your .csv file to the fields in the IPitomy Dialer. Place your mouse over the column field in the left column that best matches the field on the Right side and drag it over to the right column. Once you have matched up the fields to want to map, then click Import. That’s all there is to it. You can update the list any time by re-importing it or manually adding the contact entries. Once you have imported the contacts, your list will be populated with the contact information and the dialer will dial the numbers when you select a contact When you double click a contact, the number is dialed. Your phone will go off hook (if you are using a phone that is capable) and your number is dialed.&lt;br /&gt;
&lt;br /&gt;
=== Integration with Telify ===&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{{:Telify}}&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPitomy_Contact_Dialer&amp;diff=5115</id>
		<title>IPitomy Contact Dialer</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPitomy_Contact_Dialer&amp;diff=5115"/>
		<updated>2024-11-13T15:56:35Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;__TOC__&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Ipitomy Contact Dialer'''&lt;br /&gt;
&lt;br /&gt;
To set up the IPitomy Dialer you must first be on the latest (5.0.7-X) PBX software.&amp;lt;br/&amp;gt;&amp;lt;br/&amp;gt;Dialer requires .NET Framework 4.0 or greater. Windows 7 or newer Microsoft OS. You also need to be licensed for the applications use. This license may be obtained from Ipitomy Sales if you do not already have it. Until this license is applied to the PBX you will not have the ability to create the Feature Key.&lt;br /&gt;
&lt;br /&gt;
'''Note'''&lt;br /&gt;
If you are having issues launching the Contact Dialer and have verified your information is correct, ensure the Time on your PC is accurate.  If the time is off from that of the PBX it will give a 408 Timeout error.  You also may need to ensure that the Daylight Savings Time box is checked in Windows or you may also get the 408 Timeout error.&lt;br /&gt;
&lt;br /&gt;
The application can be downloaded from:&lt;br /&gt;
&lt;br /&gt;
[http://relay.ipitomy.com/contacts/Contacts.application Click here to download the installation file]&lt;br /&gt;
&lt;br /&gt;
If clicking it does not initiate the download, try a different browser. You can also right click, select &amp;quot;copy link address&amp;quot; then paste the link address into another browser tab.&lt;br /&gt;
&lt;br /&gt;
Once you have upgraded you will have under the Applications area of the PBX GUI an API Page.&lt;br /&gt;
&lt;br /&gt;
=== Installation Video ===&lt;br /&gt;
&lt;br /&gt;
https://www.youtube.com/watch?v=6ginRqzHcNc&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;Use the tool to create a KEY named &amp;quot;Contacts&amp;quot; (case sensitive)&lt;br /&gt;
&lt;br /&gt;
[[File:Ipitomy Dialer1.jpg|File:Ipitomy Dialer1.jpg]]&lt;br /&gt;
&lt;br /&gt;
Once finished your Key will look like this.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:IpDialer.jpg|File:IpDialer.jpg]]&lt;br /&gt;
&lt;br /&gt;
At this point you can use the guide on the following link to configure and use the dialer.&lt;br /&gt;
&lt;br /&gt;
[http://wiki.ipitomy.com/wiki/File:IPitomy_Dialer_App.pdf http://wiki.ipitomy.com/wiki/File:IPitomy_Dialer_App.pdf]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;The installation will begin. When finished it will bring you to the configuration page.&lt;br /&gt;
&lt;br /&gt;
[[File:Contact 1.jpg|File:Contact 1.jpg]]&lt;br /&gt;
&lt;br /&gt;
Enter the Extension number where indicated. The Password used is the Extension Voicemail Password. And then populate the IP of the PBX in the appropriate field. No changeto the port number is required.&lt;br /&gt;
&lt;br /&gt;
This will bring up the main page.&lt;br /&gt;
&lt;br /&gt;
[[File:Contact 2.png|File:Contact 2.png]]&lt;br /&gt;
&lt;br /&gt;
System Speed dials are present. To dial these simply click on them. The device you configured will ring. And once you pick up the call will be connecting. You can import contacts from a CSV file or from a local Outlook installation. You can also create manually the entries you desire. Enter the Extension number.&lt;br /&gt;
&lt;br /&gt;
'''&amp;amp;nbsp;IPitomy Dialer'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;&amp;lt;br/&amp;gt;'''Overview'''&amp;lt;br/&amp;gt;Many users of business phone systems call a wide variety of phone numbers. They can be from a printed sheet, a contact program or from Microsoft Outlook. When there are many calls to make and each number has to be dialed into a phone, the process becomes tedious. To ease the process of dialing each number, IPitomy has created the IPitomy Dialer. The&amp;lt;br/&amp;gt;dialer can be used in several ways:&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;1) Automatically import contacts from Microsoft Outlook. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;2) Import Contacts from a .csv (comma separated values) file. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;3) Dialing System speed dial numbers. The PBX will automatically import any system speed dial numbers. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;4) Click to dial from Mozilla Firefox. using the Telify add-on, phone numbers in web pages can be clicked to dial.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;5) Clip to Dial - Copy - by selecting a phone number in any program running on your desktop, Select the phone number by highlighting it with a mouse. Click Control C then Control D and the number is dialed. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
=== Program Setup ===&lt;br /&gt;
&lt;br /&gt;
You first need to add your extension and Pin number to the Dialer in order for the IPitomyPBX to dial from your extension. Click Settings in the upper left corner Enter your extension number in the Extension field Enter the PIN number you received for your Voice mail Box Add the IPitomy server IPaddress. You will get this from your system administrator. Click Connect Now the Dialer is setup to connect with the PBX and dial out numbers from your desktop.Before you can begin dialing, it is necessary to populate the contact dialer with contacts so you can dial them with just a click. To import Contacts from Microsoft Outlook, you must first be using Outlook 2007 or newer. Click on Advanced under the Managed Contacts tab. After clicking the Advanced tab, you will see the option to import Outlook Contacts. Click the Import Outlook Contacts tab, once this is clicked, the Dialer will import all of your contacts. It may take up to 10 minutes, depending on how many contacts you have&lt;br /&gt;
&lt;br /&gt;
=== Importing Contacts ===&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;From Outlook Click Manage Contacts Click Import Click Import Outlook ContactsThe Dialer will import all of your Outlook Contacts. Once imported they will appear ina list. just click on the item in the list you wish to dial.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Importing Contacts From a .csv File'''&amp;lt;br/&amp;gt;Click Manage Contacts Click Import Click Import from CSV file Older versions of outlook and Combinations of contacts from a .csv file contacts and outlook contacts. In order to import contacts from older versions of Outlook, it may be necessary to first export your Outlook Contacts to a .csv file. Then import the file using the .csv import function.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Using The Import From .csv Function'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;To import contacts from a .csv file, first, make sure your .CSV file has a header line. '''''The Contact dialer will ignore the first row of the .CSV'''''. It is necessary to map the fields in the contact from the .csv file to match up the fields in the IPitomy Dialer contact fields. To do this, you click on Import Contacts from CSV Then browse to the file you want to import on your PC Select the file and click Open The file will then begin the import process. It is now necessary to map the fields from your .csv file to the fields in the IPitomy Dialer. Place your mouse over the column field in the left column that best matches the field on the Right side and drag it over to the right column. Once you have matched up the fields to want to map, then click Import. That’s all there is to it. You can update the list any time by re-importing it or manually adding the contact entries. Once you have imported the contacts, your list will be populated with the contact information and the dialer will dial the numbers when you select a contact When you double click a contact, the number is dialed. Your phone will go off hook (if you are using a phone that is capable) and your number is dialed.&lt;br /&gt;
&lt;br /&gt;
=== Integration with Telify ===&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{{:Telify}}&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPitomy_Contact_Dialer&amp;diff=5114</id>
		<title>IPitomy Contact Dialer</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPitomy_Contact_Dialer&amp;diff=5114"/>
		<updated>2024-11-13T15:47:19Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: Added a detail about the .csv format.&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;__TOC__&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Ipitomy Contact Dialer'''&lt;br /&gt;
&lt;br /&gt;
To set up the IPitomy Dialer you must first be on the latest (5.0.7-X) PBX software.&amp;lt;br/&amp;gt;&amp;lt;br/&amp;gt;Dialer requires .NET Framework 4.0 or greater. Windows 7 or newer Microsoft OS. You also need to be licensed for the applications use. This license may be obtained from Ipitomy Sales if you do not already have it. Until this license is applied to the PBX you will not have the ability to create the Feature Key.&lt;br /&gt;
&lt;br /&gt;
'''Note'''&lt;br /&gt;
If you are having issues launching the Contact Dialer and have verified your information is correct, ensure the Time on your PC is accurate.  If the time is off from that of the PBX it will give a 408 Timeout error.  You also may need to ensure that the Daylight Savings Time box is checked in Windows or you may also get the 408 Timeout error.&lt;br /&gt;
&lt;br /&gt;
The application can be downloaded from:&lt;br /&gt;
&lt;br /&gt;
[http://relay.ipitomy.com/contacts/Contacts.application Click here to download the installation file]&lt;br /&gt;
&lt;br /&gt;
Once you have upgraded you will have under the Applications area of the PBX GUI an API Page.&lt;br /&gt;
&lt;br /&gt;
=== Installation Video ===&lt;br /&gt;
&lt;br /&gt;
https://www.youtube.com/watch?v=6ginRqzHcNc&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;Use the tool to create a KEY named &amp;quot;Contacts&amp;quot; (case sensitive)&lt;br /&gt;
&lt;br /&gt;
[[File:Ipitomy Dialer1.jpg|File:Ipitomy Dialer1.jpg]]&lt;br /&gt;
&lt;br /&gt;
Once finished your Key will look like this.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:IpDialer.jpg|File:IpDialer.jpg]]&lt;br /&gt;
&lt;br /&gt;
At this point you can use the guide on the following link to configure and use the dialer.&lt;br /&gt;
&lt;br /&gt;
[http://wiki.ipitomy.com/wiki/File:IPitomy_Dialer_App.pdf http://wiki.ipitomy.com/wiki/File:IPitomy_Dialer_App.pdf]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;The installation will begin. When finished it will bring you to the configuration page.&lt;br /&gt;
&lt;br /&gt;
[[File:Contact 1.jpg|File:Contact 1.jpg]]&lt;br /&gt;
&lt;br /&gt;
Enter the Extension number where indicated. The Password used is the Extension Voicemail Password. And then populate the IP of the PBX in the appropriate field. No changeto the port number is required.&lt;br /&gt;
&lt;br /&gt;
This will bring up the main page.&lt;br /&gt;
&lt;br /&gt;
[[File:Contact 2.png|File:Contact 2.png]]&lt;br /&gt;
&lt;br /&gt;
System Speed dials are present. To dial these simply click on them. The device you configured will ring. And once you pick up the call will be connecting. You can import contacts from a CSV file or from a local Outlook installation. You can also create manually the entries you desire. Enter the Extension number.&lt;br /&gt;
&lt;br /&gt;
'''&amp;amp;nbsp;IPitomy Dialer'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;&amp;lt;br/&amp;gt;'''Overview'''&amp;lt;br/&amp;gt;Many users of business phone systems call a wide variety of phone numbers. They can be from a printed sheet, a contact program or from Microsoft Outlook. When there are many calls to make and each number has to be dialed into a phone, the process becomes tedious. To ease the process of dialing each number, IPitomy has created the IPitomy Dialer. The&amp;lt;br/&amp;gt;dialer can be used in several ways:&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;1) Automatically import contacts from Microsoft Outlook. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;2) Import Contacts from a .csv (comma separated values) file. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;3) Dialing System speed dial numbers. The PBX will automatically import any system speed dial numbers. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;4) Click to dial from Mozilla Firefox. using the Telify add-on, phone numbers in web pages can be clicked to dial.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;5) Clip to Dial - Copy - by selecting a phone number in any program running on your desktop, Select the phone number by highlighting it with a mouse. Click Control C then Control D and the number is dialed. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
=== Program Setup ===&lt;br /&gt;
&lt;br /&gt;
You first need to add your extension and Pin number to the Dialer in order for the IPitomyPBX to dial from your extension. Click Settings in the upper left corner Enter your extension number in the Extension field Enter the PIN number you received for your Voice mail Box Add the IPitomy server IPaddress. You will get this from your system administrator. Click Connect Now the Dialer is setup to connect with the PBX and dial out numbers from your desktop.Before you can begin dialing, it is necessary to populate the contact dialer with contacts so you can dial them with just a click. To import Contacts from Microsoft Outlook, you must first be using Outlook 2007 or newer. Click on Advanced under the Managed Contacts tab. After clicking the Advanced tab, you will see the option to import Outlook Contacts. Click the Import Outlook Contacts tab, once this is clicked, the Dialer will import all of your contacts. It may take up to 10 minutes, depending on how many contacts you have&lt;br /&gt;
&lt;br /&gt;
=== Importing Contacts ===&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;From Outlook Click Manage Contacts Click Import Click Import Outlook ContactsThe Dialer will import all of your Outlook Contacts. Once imported they will appear ina list. just click on the item in the list you wish to dial.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Importing Contacts From a .csv File'''&amp;lt;br/&amp;gt;Click Manage Contacts Click Import Click Import from CSV file Older versions of outlook and Combinations of contacts from a .csv file contacts and outlook contacts. In order to import contacts from older versions of Outlook, it may be necessary to first export your Outlook Contacts to a .csv file. Then import the file using the .csv import function.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Using The Import From .csv Function'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;To import contacts from a .csv file, first, make sure your .CSV file has a header line. '''''The Contact dialer will ignore the first row of the .CSV'''''. It is necessary to map the fields in the contact from the .csv file to match up the fields in the IPitomy Dialer contact fields. To do this, you click on Import Contacts from CSV Then browse to the file you want to import on your PC Select the file and click Open The file will then begin the import process. It is now necessary to map the fields from your .csv file to the fields in the IPitomy Dialer. Place your mouse over the column field in the left column that best matches the field on the Right side and drag it over to the right column. Once you have matched up the fields to want to map, then click Import. That’s all there is to it. You can update the list any time by re-importing it or manually adding the contact entries. Once you have imported the contacts, your list will be populated with the contact information and the dialer will dial the numbers when you select a contact When you double click a contact, the number is dialed. Your phone will go off hook (if you are using a phone that is capable) and your number is dialed.&lt;br /&gt;
&lt;br /&gt;
=== Integration with Telify ===&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{{:Telify}}&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPitomy_Contact_Dialer&amp;diff=5112</id>
		<title>IPitomy Contact Dialer</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPitomy_Contact_Dialer&amp;diff=5112"/>
		<updated>2024-11-07T19:28:29Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: Re-added the download link&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;__TOC__&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Ipitomy Contact Dialer'''&lt;br /&gt;
&lt;br /&gt;
To set up the IPitomy Dialer you must first be on the latest (5.0.7-X) PBX software.&amp;lt;br/&amp;gt;&amp;lt;br/&amp;gt;Dialer requires .NET Framework 4.0 or greater. Windows 7 or newer Microsoft OS. You also need to be licensed for the applications use. This license may be obtained from Ipitomy Sales if you do not already have it. Until this license is applied to the PBX you will not have the ability to create the Feature Key.&lt;br /&gt;
&lt;br /&gt;
'''Note'''&lt;br /&gt;
If you are having issues launching the Contact Dialer and have verified your information is correct, ensure the Time on your PC is accurate.  If the time is off from that of the PBX it will give a 408 Timeout error.  You also may need to ensure that the Daylight Savings Time box is checked in Windows or you may also get the 408 Timeout error.&lt;br /&gt;
&lt;br /&gt;
The application can be downloaded from:&lt;br /&gt;
&lt;br /&gt;
[http://relay.ipitomy.com/contacts/Contacts.application Click here to download the installation file]&lt;br /&gt;
&lt;br /&gt;
Once you have upgraded you will have under the Applications area of the PBX GUI an API Page.&lt;br /&gt;
&lt;br /&gt;
=== Installation Video ===&lt;br /&gt;
&lt;br /&gt;
https://www.youtube.com/watch?v=6ginRqzHcNc&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;Use the tool to create a KEY named &amp;quot;Contacts&amp;quot; (case sensitive)&lt;br /&gt;
&lt;br /&gt;
[[File:Ipitomy Dialer1.jpg|File:Ipitomy Dialer1.jpg]]&lt;br /&gt;
&lt;br /&gt;
Once finished your Key will look like this.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;[[File:IpDialer.jpg|File:IpDialer.jpg]]&lt;br /&gt;
&lt;br /&gt;
At this point you can use the guide on the following link to configure and use the dialer.&lt;br /&gt;
&lt;br /&gt;
[http://wiki.ipitomy.com/wiki/File:IPitomy_Dialer_App.pdf http://wiki.ipitomy.com/wiki/File:IPitomy_Dialer_App.pdf]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;The installation will begin. When finished it will bring you to the configuration page.&lt;br /&gt;
&lt;br /&gt;
[[File:Contact 1.jpg|File:Contact 1.jpg]]&lt;br /&gt;
&lt;br /&gt;
Enter the Extension number where indicated. The Password used is the Extension Voicemail Password. And then populate the IP of the PBX in the appropriate field. No changeto the port number is required.&lt;br /&gt;
&lt;br /&gt;
This will bring up the main page.&lt;br /&gt;
&lt;br /&gt;
[[File:Contact 2.png|File:Contact 2.png]]&lt;br /&gt;
&lt;br /&gt;
System Speed dials are present. To dial these simply click on them. The device you configured will ring. And once you pick up the call will be connecting. You can import contacts from a CSV file or from a local Outlook installation. You can also create manually the entries you desire. Enter the Extension number.&lt;br /&gt;
&lt;br /&gt;
'''&amp;amp;nbsp;IPitomy Dialer'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;&amp;lt;br/&amp;gt;'''Overview'''&amp;lt;br/&amp;gt;Many users of business phone systems call a wide variety of phone numbers. They can be from a printed sheet, a contact program or from Microsoft Outlook. When there are many calls to make and each number has to be dialed into a phone, the process becomes tedious. To ease the process of dialing each number, IPitomy has created the IPitomy Dialer. The&amp;lt;br/&amp;gt;dialer can be used in several ways:&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;1) Automatically import contacts from Microsoft Outlook. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;2) Import Contacts from a .csv (comma separated values) file. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;3) Dialing System speed dial numbers. The PBX will automatically import any system speed dial numbers. Once the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;4) Click to dial from Mozilla Firefox. using the Telify add-on, phone numbers in web pages can be clicked to dial.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;5) Clip to Dial - Copy - by selecting a phone number in any program running on your desktop, Select the phone number by highlighting it with a mouse. Click Control C then Control D and the number is dialed. Your phone will go off hook and the call will be put through without the need to enter any digits.&lt;br /&gt;
&lt;br /&gt;
=== Program Setup ===&lt;br /&gt;
&lt;br /&gt;
You first need to add your extension and Pin number to the Dialer in order for the IPitomyPBX to dial from your extension. Click Settings in the upper left corner Enter your extension number in the Extension field Enter the PIN number you received for your Voice mail Box Add the IPitomy server IPaddress. You will get this from your system administrator. Click Connect Now the Dialer is setup to connect with the PBX and dial out numbers from your desktop.Before you can begin dialing, it is necessary to populate the contact dialer with contacts so you can dial them with just a click. To import Contacts from Microsoft Outlook, you must first be using Outlook 2007 or newer. Click on Advanced under the Managed Contacts tab. After clicking the Advanced tab, you will see the option to import Outlook Contacts. Click the Import Outlook Contacts tab, once this is clicked, the Dialer will import all of your contacts. It may take up to 10 minutes, depending on how many contacts you have&lt;br /&gt;
&lt;br /&gt;
=== Importing Contacts ===&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;From Outlook Click Manage Contacts Click Import Click Import Outlook ContactsThe Dialer will import all of your Outlook Contacts. Once imported they will appear ina list. just click on the item in the list you wish to dial.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Importing Contacts From a .csv File'''&amp;lt;br/&amp;gt;Click Manage Contacts Click Import Click Import from CSV file Older versions of outlook and Combinations of contacts from a .csv file contacts and outlook contacts. In order to import contacts from older versions of Outlook, it may be necessary to first export your Outlook Contacts to a .csv file. Then import the file using the .csv import function.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;'''Using The Import From .csv Function'''&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;To import contacts from a .csv file, it is necessary to map the fields in the contact from the .csv file to match up the fields in the IPitomy Dialer contact fields. To do this, you click on Import Contacts from CSV Then browse to the file you want to import on your PC Select the file and click Open The file will then begin the import process. It is now necessary to map the fields from your .csv file to the fields in the IPitomy Dialer. Place your mouse over the column field in the left column that best matches the field on the Right side and drag it over to the right column. Once you have matched up the fields to want to map, then click Import. That’s all there is to it. You can update the list any time by re-importing it or manually adding the contact entries. Once you have imported the contacts, your list will be populated with the contact information and the dialer will dial the numbers when you select a contact When you double click a contact, the number is dialed. Your phone will go off hook (if you are using a phone that is capable) and your number is dialed.&lt;br /&gt;
&lt;br /&gt;
=== Integration with Telify ===&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{{:Telify}}&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=Grandstream_410X_Setup&amp;diff=5109</id>
		<title>Grandstream 410X Setup</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=Grandstream_410X_Setup&amp;diff=5109"/>
		<updated>2024-07-19T18:18:55Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Grandstream GXW 410X Gateway and HT841 FXO FXS Gateway */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;== Grandstream GXW 410X Gateway and HT841 FXO FXS Gateway ==&lt;br /&gt;
&lt;br /&gt;
If you are installing and HT841 gateway, you will program it the same as the GXW 410X below with 2 added settings.  &lt;br /&gt;
&lt;br /&gt;
# In the SIP provider page you will change the Port to Custom and make it 6062.&lt;br /&gt;
# In the HT841 you will wet the FXO Port settings for the Hunt Group to port 1. Active, and the rest of the ports to 1.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:HT841.png|frameless|600x600px]]&lt;br /&gt;
&lt;br /&gt;
[[File:HT841 ports.png|none|thumb|600x600px]]&lt;br /&gt;
[[File:Ht881.png|left|frameless|602x602px]]&lt;br /&gt;
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[[File:Ht881-2.png|left|thumb|600x600px]]&lt;br /&gt;
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&lt;br /&gt;
Description&lt;br /&gt;
&lt;br /&gt;
The following configuration guide is for use setting up the Grandstream GXW410X gateways with analog circuits.&lt;br /&gt;
&lt;br /&gt;
NOTE: This guide is based upon the most current software from Grandstream. However if you have older firmware this is a link to the older guide which documents and covers the older firmware. &amp;amp;nbsp;The 4104 and 4108 use the exact same firmware as each other and have the same interface.&lt;br /&gt;
&lt;br /&gt;
[http://wiki.ipitomy.com/images/5/57/Tech_Bulletin_2011-008_Grandstream_Configuration_Guide.pdf http://wiki.ipitomy.com/images/5/57/Tech_Bulletin_2011-008_Grandstream_Configuration_Guide.pdf]&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream1.JPG|File:Gstream1.JPG]]&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Connections ===&lt;br /&gt;
&lt;br /&gt;
1. Connect each analog circuit to the desired FXO port, as above&lt;br /&gt;
&lt;br /&gt;
2. Connect the Grandstream WAN port to an available LAN port of the network switch/router being used on site.&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream2.JPG|File:Gstream2.JPG]]&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Power Up and Login ===&lt;br /&gt;
&lt;br /&gt;
1. Power up the unit and identify its assigned IP address. (Typically assigned from the DCHP server of the host router.)&lt;br /&gt;
&lt;br /&gt;
2. Use your browser to access the Grandstream by inputting the IP Address assigned to it. The IP Address assigned by your router via DHCP can be discovered several ways – the easiest of which is likely by accessing the router’s connected devices page and finding it listed there.&lt;br /&gt;
&lt;br /&gt;
3. When the Grandstream page is accessed, input the password (admin at default) and navigate to the pages below making the changes as defined.&lt;br /&gt;
&lt;br /&gt;
=== IMPORTANT: Procedure Line Setup&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
On first login to the Grandstream, you will be brought to the following page.&lt;br /&gt;
&lt;br /&gt;
[[File:GrandstreamCPT.jpg|File:GrandstreamCPT.jpg]]&lt;br /&gt;
&lt;br /&gt;
In ALL new installs once you have the device on the Network and lines connected, you should run each of these tests. The first is for Call quality and is MOST important. As this test matches line impedance which if it is in variance can cause echo and poor calls. The second allows the device to learn Call Progress Tones (CPT) and is also very important so that the Unit can learn disconnect and busy tones on the lines. NOTE: The Grandstream can learn a Disconnect Supervision Current drop and a busy tone for disconnect. Lastly is the test for CID. This is also important and additionally can detect CID presence on the line. For proper performance of the Grandstream Device these tests SHOULD NOT be skipped.&lt;br /&gt;
&lt;br /&gt;
NOTE: These tests should be performed at the site and with the lines that the unit is intended to be in service upon.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuration&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
1. Navigate to the Networks Basic Setting page and input the data as in the example.&lt;br /&gt;
&lt;br /&gt;
[[File:SIP3.jpg|File:SIP3.jpg]]&amp;lt;br/&amp;gt;2. Assign a Static IP Address. The device must be found by the IPBX regardless of incidental changes and network adjustments. For this reason its best to change the IP Address to Static and assign an address that is out of range of those assigned for DCHP subscription. (E.g. if the router will assign DCHP Addresses from 192.168.1.1 ~ 192.168.1.50 you should select an IP Address out of this range ...192.168.1.200 would work unless it is being used elsewhere.)&amp;lt;br/&amp;gt;3. Use the other information provided by the DCHP assignment process in the remaining data fields; Subnet is usually 255.255.255.0. The Default Router Address must be that of the router the same one that assigns DCHP IP Addresses. DNS should also be the router since it will direct traffic.&amp;lt;br/&amp;gt;4. Click the save button. This saves information on this page before moving on.&amp;lt;br/&amp;gt;5. Navigate to the FXO Lines page. Then Select Dialing on the left.&amp;lt;br/&amp;gt;6. Change the Stage Method(1/2)&amp;lt;br/&amp;gt;Ch1-4:1;&lt;br /&gt;
&lt;br /&gt;
[[File:Gxw410x stagemethod.PNG|File:Gxw410x stagemethod.PNG]]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;7. Navigate to Settings and then select Channels Settings on the left. Program DTMF Methods to ch1-4:2;.  Program the Unconditional Call Forward to VOIP: to include the DID (Direct Inward Dial) number(s) that are to be routed.This routing is accomplished by Profiles 1, 2 &amp;amp; 3. Usually only one is necessary.This data field is the routing of the calls that are received on this FXO circuit (It is recomended to use our example on this line which is&amp;lt;br/&amp;gt;ch1:1111;ch2:2222;ch3:3333;ch4:4444;&amp;lt;br/&amp;gt;This will create &amp;quot;virtual DIDs&amp;quot; of 1111, 2222, etc, that allow you to enter DIDs in the PBX under the SIP Provider so you can route calls to different locations based on what line they came in on.&amp;amp;nbsp; We have programmed the Grandstream to route calls that have been received to the SIP Server using Profile 1.&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream3.JPG|File:Gstream3.JPG]]&lt;br /&gt;
&lt;br /&gt;
8.Click the save button.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuration Profile ===&lt;br /&gt;
&lt;br /&gt;
[[File:SIP.jpg|File:SIP.jpg]]&lt;br /&gt;
&lt;br /&gt;
1. Navigate to the Accounts=&amp;gt;Account 1=&amp;gt;General Settings page.&amp;lt;br/&amp;gt;2. Here the SIP Server must be programmed. Set this to be the IP Address of the IPBX. In our example, the address is 192.168.2.153.&amp;lt;br/&amp;gt;3. Also be sure that the SIP Registration field is set to No.&amp;lt;br/&amp;gt;4.Click the save button.&amp;lt;br/&amp;gt;5.The Grandstream configuration is now complete. However if you wish to make changes to the dial plan allowed digits you must also program that information.&amp;lt;br/&amp;gt;Note: At default the Grandstream allows only digits 0-9 to be sent to the connected PSTN circuit. If you want to use PSTN features like call forward you will need to change the Outgoing Dial Plan field to;&amp;lt;br/&amp;gt;{x+ | [x*]+}&lt;br /&gt;
&lt;br /&gt;
[[File:Sip2.jpg|File:Sip2.jpg]]&amp;lt;br/&amp;gt;to be able to send a * to outside IVR's&amp;lt;br/&amp;gt;6. When programming is complete in the Grandstream, click submit will commit the changes saved thus far to memory and make them operational. Continue to Procedure Configuring the IPitomy IP PBX&lt;br /&gt;
&lt;br /&gt;
[[File:Gstream5.JPG|File:Gstream5.JPG]]&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuring the IPitomy IP PBX for Grandstream GXW 410x ===&lt;br /&gt;
&lt;br /&gt;
1.In the IPitomy IPBX set the fields as you see them below... using the Static IP Address assigned to the Procedure—Configuration step previously completed.. (In our example we assigned the Grandstream an IP Address of 192.168.2.9. This becomes the ―Host)&lt;br /&gt;
&lt;br /&gt;
2. Click “Save Changes”&lt;br /&gt;
&lt;br /&gt;
3. Then click on theApply Changes (upper right) to make these settings operational in the Ipitomy IPBX.&lt;br /&gt;
&lt;br /&gt;
4.Test the operation. Make a call into each of the Grandstream ports that have circuits and assure that they are being routed as defined in Call Routing—Incoming.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;5.Test the operation. Make a call at an Ipitomy extension using a calling pattern as defined in Call Routing Outgoing to assure that the call that should be placed over the Grandstream ports are placed.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configuring Optional Outbound Routing Methods ===&lt;br /&gt;
&lt;br /&gt;
The Grandstream allows for two methods of outbound dialing at the same time.&lt;br /&gt;
&lt;br /&gt;
Round-Robin (Linear Hunt)&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;If you add the trunk to an outbound route in the PBX, the Grandstream will follow the Round-Robin rules, which are found on the Channels page. By default it will start at Line&lt;br /&gt;
&lt;br /&gt;
1, and move on until it finds an available channel to dial outbound.If you add the trunk to an outbound route and configure Prefix Digits to 99X where X is the port on the Grandstream you wish to use when placing this call (e.g. 991 is line 1). By doing this you can configure the calls to route out a particular line, or a different order if you add the trunk multiple times and prefix accordingly (992, 994, 991, 993, etc). The code of 99 can be changed on the Channels page in the Grandstream.&lt;br /&gt;
&lt;br /&gt;
1. Navigate to the Channels page&lt;br /&gt;
&lt;br /&gt;
2. Go to the Port Scheduling Schema (Voip-&amp;gt;PSTN) section and input the code or codes that you wish (99x)&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Configure Multiple Line Groups on Single Grandstream ===&lt;br /&gt;
&lt;br /&gt;
'''1 -''' In order to put the lines into separate outbound groups you will need to configure each round robin group under FXO Lines =&amp;gt; Dialing.  In this example lines 1-3 are in the first group and line 4 is standalone.&lt;br /&gt;
&lt;br /&gt;
[[File:GS RR Separate.PNG|File:GS RR Separate.PNG]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''2 -''' After setting up the round robin groups you now have to configure the local SIP port for the groups.  In this example channels 1-3 are using 5060 and incrementing by 2 (this is default behavior for multiple channels in a group, any call sent to 5060 will use the first available line in the group), channel 4 is alone listening on port 5160.&lt;br /&gt;
&lt;br /&gt;
[[File:GS RR Ports.PNG]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''3 -''' After configuring the local SIP port you now need to configure the account SIP User ID, this will allow the PBX to differentiate between each line group on incoming calls.  Lines 1-3 will give a user ID of 0001, and line 4 will give a user ID of 0002.&lt;br /&gt;
&lt;br /&gt;
[[File:GS RR Account.PNG]]&lt;br /&gt;
&lt;br /&gt;
'''4 -''' On the PBX side we will need to create two trunks, both pointing to the same IP address of the Grandstream, but each will have a different user.  '''Be absolutely certain you configure the trunk with Insecure = No''', this setting will tell the PBX to consider the user when taking the call.  Notice that on the 2nd trunk we've added the correct port number for line 4 and the correct User associated with line 4.  These two settings are what make the 2nd trunk use line 4, the first trunk is using the default port of 5060 and therefore only needs the user to distinguish incoming calls.&lt;br /&gt;
&lt;br /&gt;
[[File:PBX RR Trunk.PNG]]&lt;br /&gt;
&lt;br /&gt;
Outgoing calls will now be sent to port 5060 on the Grandstream to use lines 1-3, calls sent to 5160 will use line 4.  Incoming calls on lines 1-3 will hit the PBX with the user of 0001 which will select the first trunk, incoming calls on line 4 will hit the PBX with user 0002 selecting the 2nd trunk.&lt;br /&gt;
&lt;br /&gt;
== Remote Grandstream Configuration ==&lt;br /&gt;
&lt;br /&gt;
In order to get a Grandstream 410x working as a remote device, you must make a few changes from the standard, local configuration. Use this guide in conjunction with the standard guide.&lt;br /&gt;
&lt;br /&gt;
=== Grandstream Changes ===&lt;br /&gt;
&lt;br /&gt;
#On the Accounts=&amp;gt;Account X=&amp;gt;SIP Settings page, set SIP Registration to YES&lt;br /&gt;
#Under the User Accounts page, configure one channel with login credentials:&lt;br /&gt;
#Set the SIP server in the Profile to the PBX Public IP.&lt;br /&gt;
&lt;br /&gt;
[[File:Gxw410x remotesettings.PNG|File:Gxw410x remotesettings.PNG]]&lt;br /&gt;
&lt;br /&gt;
=== PBX SIP Provider Changes&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
#Instead of entering an IP address for Host, set to “dynamic”&lt;br /&gt;
#Set Register and Authentication to YES&lt;br /&gt;
#Set Username and Name to the value entered as SIP User ID and Authenticate ID in the grandstream&lt;br /&gt;
#Set Secret to the value entered for Authen Password in the grandstream&lt;br /&gt;
&lt;br /&gt;
Note: If the grandstream is not the only device at the remote site, then all remote SIP devices will need to have Can Reinvite set to YES.&lt;br /&gt;
&lt;br /&gt;
== Troubleshooting ==&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Outbound Dialing ===&lt;br /&gt;
&lt;br /&gt;
If you having trouble dialing outbound make the following changes on the FXO Lines page:&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;1. Tweak the Disconnect Threshold from 100 to 300ms.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;2. Tweak the Minimum Delay Before Dialing Out from 500 to 750ms.&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Call Quality ===&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;If you are having call quality issues try the following changes on the Channels page:&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;1.Set Silence Suppression from YES to NO.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;2. Set Echo Cancellation from YES to NO&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Call Volume ===&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;If you are having issue with call volume try the following changes on the Channels page:&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;1.Increase/Decrease Tx to PSTN Audio Gain by increments of 3 for issues with external party volume.&lt;br /&gt;
&lt;br /&gt;
&amp;amp;nbsp;2. Increase/Decrease Rx from PSTN Audio Gain by increments of 3 for issues with internal party volume&lt;br /&gt;
&lt;br /&gt;
&amp;lt;span style=&amp;quot;font-size:large&amp;quot;&amp;gt;'''&amp;amp;nbsp;Procedure—Troubleshooting—Call Buzzing Noise'''&amp;lt;/span&amp;gt;&lt;br /&gt;
&lt;br /&gt;
If you are having issues with a buzz heard prior to a Menu prompt; try upgrading the Grandstream firmware:&lt;br /&gt;
&lt;br /&gt;
1. Navigate to Advanced Settings page&lt;br /&gt;
&lt;br /&gt;
2. Ensure HTTP is selected for the method to upgrade&lt;br /&gt;
&lt;br /&gt;
3. Set Firmware Server Path: to firmware.grandstream.com&lt;br /&gt;
&lt;br /&gt;
4. Set Automatic Upgrade to YES&lt;br /&gt;
&lt;br /&gt;
5. Set Allow DHCP Option 66 to override server to No Click Update at the bottom of the page&lt;br /&gt;
&lt;br /&gt;
6. Click Reboot The upgrade may take as long as 20min when done through the internet, so allow plenty of time for this. While upgrading the LED will blink. When the LED returns to normal, the device has completed its upgrade&lt;br /&gt;
&lt;br /&gt;
=== Procedure—Troubleshooting—Restore Factory Default ===&lt;br /&gt;
&lt;br /&gt;
To Restore Factory Defaults:&lt;br /&gt;
&lt;br /&gt;
1. While powered up, hold the recessed Reset button in for 7+ seconds&lt;br /&gt;
&lt;br /&gt;
'''&amp;lt;span style=&amp;quot;font-size:large&amp;quot;&amp;gt;[[File:Gstream6.JPG|File:Gstream6.JPG]]&amp;amp;nbsp;&amp;lt;/span&amp;gt;'''&lt;br /&gt;
&lt;br /&gt;
'''&amp;lt;span style=&amp;quot;font-size:large&amp;quot;&amp;gt;Procedure Setup of QOS&amp;lt;/span&amp;gt;'''&lt;br /&gt;
&lt;br /&gt;
1. Access the IP of the Grandstream.&lt;br /&gt;
&lt;br /&gt;
2. Navigate in the Grandstream Web Admin to Networks – Advanced Settings&lt;br /&gt;
&lt;br /&gt;
3. Set the Layer 3 Diff Serv Value to 24 Which is CS3 or in Binary 011000&lt;br /&gt;
&lt;br /&gt;
4. Submit your changes and reboot&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br/&amp;gt;{{:Grandstream_FXO_FAQ}}&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
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		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=File:Ht881-2.png&amp;diff=5108"/>
		<updated>2024-07-19T18:18:15Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;jw&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=File:Ht881.png&amp;diff=5107</id>
		<title>File:Ht881.png</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=File:Ht881.png&amp;diff=5107"/>
		<updated>2024-07-19T18:16:34Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;jw&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPPBX_IMM_SystemAdminMonitoring&amp;diff=5105</id>
		<title>IPPBX IMM SystemAdminMonitoring</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPPBX_IMM_SystemAdminMonitoring&amp;diff=5105"/>
		<updated>2024-06-10T21:39:11Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Monitoring */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;__NOTITLE__&lt;br /&gt;
{{IP_PBX_Manual|sortkey=Monitoring}}&lt;br /&gt;
== Monitoring ==&lt;br /&gt;
This page is located under Reporting -&amp;gt; Monitoring in the PBX.  This page gives a snapshot of the status of your SIP network. Extensions, SIP Providers, Branch Offices, and active calls will display. Entries in Green are connected, entries in Red are unable to be reached, and entires in Grey are not present. At the bottom of the page, active calls will display on a channel by channel basis. The page will refresh every 10 seconds.&lt;br /&gt;
[[File:monitoringpage.png|center]]&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IP_PBX_Manual_PBXSetup_Voicemail&amp;diff=5102</id>
		<title>IP PBX Manual PBXSetup Voicemail</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IP_PBX_Manual_PBXSetup_Voicemail&amp;diff=5102"/>
		<updated>2024-05-16T15:03:56Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;{{IP_PBX_Manual|sortkey=Voicemail Global}}&lt;br /&gt;
&lt;br /&gt;
== Voicemail Setup ==&lt;br /&gt;
&lt;br /&gt;
This page is used to configure global voicemail box settings, interact with the voicemail archive, as well as configuring the PBX to send out email notifications.&lt;br /&gt;
&lt;br /&gt;
=== General Settings Section ===&lt;br /&gt;
&lt;br /&gt;
Used to configure global mailbox settings that are not found elsewhere in the PBX.&lt;br /&gt;
&lt;br /&gt;
[[File:Vmailgensettings.png|center|Vmailgensettings.png]]&lt;br /&gt;
&lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Sections/Fields'''&amp;lt;/center&amp;gt;&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Description'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Max Number of Messages'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Maximum number of messages allowed for a voicemail box. Default value is '''100.'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Max Message Length '''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Maximum length of a message in seconds. Default value is '''180.'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Min Message Length'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Minimum length of a message in seconds. Default value is '''3.'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Max Greeting Length'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Maximum length of a greeting in seconds. Default value is '''60.'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Max Seconds of Silence'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Maximum seconds of silence before the message is considered complete. Set this to zero for an infinite time period. Default value is '''10.'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Silence Threshold'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | When using the maximum Seconds of Silence setting, it is sometimes necessary to adjust the silence detection threshold to eliminate false triggering on background noise. The higher the number, the more background noise is needed to break the silence. Default value is '''128.'''&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
==== Set General Voicemail Settings&amp;lt;br/&amp;gt; ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup-&amp;gt;Voicemail Setup '''page. Scroll to find the '''General Settings '''section.&lt;br /&gt;
#Set the General Voicemail parameters base on your business. In most scenarios, the default settings are fine.&lt;br /&gt;
#Click the '''Save''' button to save the settings.&lt;br /&gt;
#Click on the '''Apply Changes''' link at the top of the page to save the information and commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
=== Voicemail Menu Section ===&lt;br /&gt;
&lt;br /&gt;
Configure settings that an individual voicemail box can follow globally.&lt;br /&gt;
&lt;br /&gt;
[[File:Vmailmenu.png|center|Vmailmenu.png]]&lt;br /&gt;
&lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Sections/Fields'''&amp;lt;/center&amp;gt;&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Description'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Play Envelope Message'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Play the envelope message (date/time) before playing the voicemail message. Default value is '''Yes.'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Say Caller ID'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Play the Caller ID information prior to the message, if available. Default value is '''Yes.'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Skip ms on playback'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | This setting defines an interval in milliseconds to use when skipping forward or reverse while a voicemail message is being played. Default value is '''3000.'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Max Failed Login Attempts'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | The number of retries a user has to enter voicemail passwords before the PBX will disconnect the user. Default value is '''3.'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''On Delete, play next msg.'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Defines if the PBX will automatically play the next message after deleting a voicemail message. Default value is '''Yes.'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow Review Mode'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Global setting that defines if callers are able to listen to a message they left, and then decide if they want to re-record it or leave it as it is. Default value is '''No.'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Allow Operator'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Global setting in regards to if a voicemail box allows callers to press “'''0'''” to reach the System Operator&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
==== Set Voicemail Menu Options ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup-&amp;gt;Voicemail Setup '''page, locate the '''Voicemail Menu '''section.&lt;br /&gt;
#Set the '''Voicemail''' '''E-mail''' parameters base on your business requirements. Typically the default setting will work.&lt;br /&gt;
#Click the [[File:Savechanges.png]] button to save the settings.&lt;br /&gt;
#Click on the '''Apply Changes''' link at the top of the page to save the information and commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
=== E-mail Settings Section&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
This section is used to define the email address and server the PBX will use to send out Unified Messaging and notification emails. Unified Messaging allows a user to receive emails whenever they receive new voicemail messages. If the PC where you check your email has the capacity to play .wav files, the you will be notified when an email is received, and will be able to listen to the new message directly from the email. Notification emails are sent out for features like Log Watch/Ban. [[File:Vmailemailsett.png|center|Vmailemailsett.png]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Sections/Fields'''&amp;lt;/center&amp;gt;&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Description'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Voicemail as Attachment'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Defines if new voicemail messages will be attached and set in an email.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''From Address'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | The email address to be used in the From address of the voicemail message. . Note: Most providers require that this is set to the address of the account used for sending the messages.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Voicemail Server'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allows voicemail services to be provided by an external server. It is not recommend to use the “Loc(local)” setting. This setting will be removed in a future release.&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Server Address'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | This is the mail server address. Enter either a fully qualified host name or an IP address.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''SSL Support'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Enable if the email server uses SSL encryption.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Server Port'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | If the email server uses a port other than 25, define that port here.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Authentication Required'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Defines if the email server requires a username and password to gain access and send messages.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''User Name'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | The user name associated with the use of the external email server.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Password'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | The external email server password.&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
==== Set E-mail Settings ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup-&amp;gt;Voicemail Setup '''page, scroll to find the '''Email Settings '''section.&lt;br /&gt;
#Set the '''Voicemail''' '''E-mail''' parameters base on your business requirements or what is recommended by IPitomy.&lt;br /&gt;
#Click the '''Save''' button to save the settings.&lt;br /&gt;
#Click on the '''Apply Changes''' link at the top of the page to save the information and commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
==== Test Settings Button ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup'''-&amp;gt;'''Voicemail Setup '''page, locate the '''E-mail Settings '''section.&lt;br /&gt;
#Located at the button of the '''E-mail Settings''' section is the '''Test Settings''' button. Click on this button to generate a test of the email settings entered.&lt;br /&gt;
#The “'''Email to send a Test message to:'''” window appears. Enter the email address that you want the test message sent to then click the '''OK''' button.&lt;br /&gt;
#If the email setting and address is valid, you will receive a “'''SENDEMAIL Successful!'''” message at the button of the Test Settings button. The email recipient that was entered will also receive an email with a subject titled “'''Voicemail Server Test'''”.&lt;br /&gt;
#If the email setting test failed, you will receive an “'''Error!'''” message at the button of the '''Test Settings''' button. Check the message for hints as to what caused the failure (ie. SSL enabled but not needed).&lt;br /&gt;
#Reconfigured the email parameters per the Set Email Settings steps above, then test again.&lt;br /&gt;
&lt;br /&gt;
Refer to the Extension Settings Section for information on configuring email addresses for Unified Messaging. Refer to the PBX Admin''''''General Section for information on configuring email addresses for Log Watch/Ban.&lt;br /&gt;
&lt;br /&gt;
=== Voicemail Archive Section ===&lt;br /&gt;
&lt;br /&gt;
Use this section to view which mailboxes currently have voicemail messages, and to download, upload, or erase the voicemail messages archived on the PBX.&lt;br /&gt;
&lt;br /&gt;
[[File:Vmailarchive.png|center|Vmailarchive.png]]&lt;br /&gt;
&lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Sections/Fields'''&amp;lt;/center&amp;gt;&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Description'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Size'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Displays the amount of space being used on the PBX for voicemail messages (Envelopes), greetings (Recordings), and the total.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''List Mailboxes Using Space'''&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Clicking on this box will display the list of mailboxes and the number of voice messages stored for each.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Download Button'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | This allows you to download a backup of all user voicemail messages, message envelopes and greeting files to your computer.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Erase Button'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Deletes all user voicemail data on the PBX, including personal greetings.&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''Upload Button'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #000000;  border-bottom:0.0069in solid #000000;  border-left:0.0069in solid #000000;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #000000;  border-bottom:0.0069in solid #000000;  border-left:none;  border-right:0.0069in solid #000000;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''IMPORTANT: Uploading a voicemail archive will overwrite existing voicemail files on the PBX.'''&amp;lt;br/&amp;gt;&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
Used to upload a previously downloaded voicemail archive.&lt;br /&gt;
&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
==== Download Voicemail Archive Settings&amp;lt;br/&amp;gt; ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup'''-&amp;gt;'''Voicemail Setup '''page, locate the '''Voicemail Archive '''section.&lt;br /&gt;
#Click on the '''Download''' button. Select '''Save File''' if prompted by your browser.&lt;br /&gt;
#Define the location you wish to save the file if prompted by your browser.&lt;br /&gt;
&lt;br /&gt;
==== Erase Voicemail Archive Settings&amp;lt;br/&amp;gt; ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup'''-&amp;gt;'''Voicemail Setup '''page, locate the '''Voicemail Archive '''section.&lt;br /&gt;
#Click on the '''Erase''' button. The warning message “This will delete all Voicemail recordings and user greeting. If you are sure you want to do this, click OK.”&lt;br /&gt;
#Click the '''OK''' button to continue. The system performs the erase and returns you to the '''Voicemail Setup''' page.&lt;br /&gt;
&lt;br /&gt;
==== Upload Voicemail Archive Settings ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup-&amp;gt;Voicemail Setup '''page, locate the '''Voicemail Archive '''section.&lt;br /&gt;
#Click on the '''Browse''' button to search for the file location of the file you want to upload.&lt;br /&gt;
#Click on the file that you want to upload. The file directory will appear in the box next to the '''Browse''' button.&lt;br /&gt;
#Click on the '''Upload '''button to initiate the upload process. You should receive a message stating that the upload was “'''Successful'''”. If the upload process failed you will receive an “'''Error'''” message indicating what failed during the process.&lt;br /&gt;
#Click the [[File:Savechanges.png]] button.&lt;br /&gt;
#Click on the '''Apply Changes''' link at the top of the page to save the information and commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
==== View Voicemail Listing (Usage Space) ====&lt;br /&gt;
&lt;br /&gt;
[[File:Vmaillisting.png|center|Vmaillisting.png]] '''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup-&amp;gt;Voicemail Setup '''page, locate the '''Voicemail Archive '''section.&lt;br /&gt;
#Click the '''List Mailboxes Using Space''' link. A listing of the Mailboxes (extensions) and number of messages for each mailbox appears.&lt;br /&gt;
#Click on the '''Hide Mailboxes Using Space''' link to close the listing.&lt;br /&gt;
&lt;br /&gt;
=== Using a Gmail Account ===&lt;br /&gt;
&lt;br /&gt;
Many times the email account provided by the end user won't work, or is giving difficulty. Its been our experience its best to not waste time trying to figure out something you don't have control over. In these cases, its advisable to create a gmail account and use that for Unified Messaging. Below is an overview of how this would be configured.&lt;br /&gt;
&lt;br /&gt;
*From Address: &amp;amp;lt;gmail email address&amp;amp;gt;&lt;br /&gt;
*Voicemail Server: Ext&lt;br /&gt;
*Server Address: smtp.gmail.com&lt;br /&gt;
*SSL Support: Yes&lt;br /&gt;
*Server Port: 465&lt;br /&gt;
*Authentication Required: Yes&lt;br /&gt;
*User Name: &amp;amp;lt;gmail email address&amp;amp;gt;&lt;br /&gt;
*Password: &amp;amp;lt;gmail password&amp;amp;gt;&lt;br /&gt;
&lt;br /&gt;
NOTE: We have received reports from our dealers in the field that you may need to enable a setting in your GMail account for &amp;quot;Access for Less Secure Apps&amp;quot; to get things working.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
If you are still encountering difficulty when using a gmail account you can access gmail settings using these links:&lt;br /&gt;
&amp;lt;div&amp;gt;1) Login to your gmail account.&amp;lt;/div&amp;gt;&amp;lt;div&amp;gt;&amp;lt;br/&amp;gt;&amp;lt;/div&amp;gt;&amp;lt;div&amp;gt;2) Go to https://www.google.com/settings/security/lesssecureapps and Turn On this feature.&amp;lt;/div&amp;gt;&amp;lt;div&amp;gt;&amp;lt;br/&amp;gt;&amp;lt;/div&amp;gt;&amp;lt;div&amp;gt;3) Go to https://accounts.google.com/DisplayUnlockCaptcha and click Continue.[[File:Less.png|none|frame]]&lt;br /&gt;
&lt;br /&gt;
=== '''Gmail less secure app fix:''' ===&lt;br /&gt;
First you will need to sign into the google account that will be responsible for sending the emails. &lt;br /&gt;
&lt;br /&gt;
= Turn on 2-Step Verification =&lt;br /&gt;
With 2-Step Verification (also known as two-factor authentication), you add an extra layer of security to your account in case your password is stolen. After you set up 2-Step Verification, you’ll sign in to your account in two steps using:&lt;br /&gt;
&lt;br /&gt;
* Something you know, like your password&lt;br /&gt;
* Something you have, like your phone&lt;br /&gt;
&lt;br /&gt;
== Activate 2-Step Verification ==&lt;br /&gt;
&lt;br /&gt;
# Open your Google Account.  Go To  accounts.google.com&lt;br /&gt;
# Click on your leter icon in the top right corner.&lt;br /&gt;
# Then, click on Manage Your Google Account&lt;br /&gt;
# In the navigation panel, select Security.&lt;br /&gt;
# Under “Signing in to Google,” select 2-Step Verification  Get started.&lt;br /&gt;
# Follow the on-screen steps.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Once you are logged in go to the account settings and search app passwords. Should be the first result to populate.[[File:Google settings.png|none|frame]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Then you will need to select the option for Email application with device type of other so we can give our password a custom Identifier.  Select &amp;quot;Other&amp;quot; and give it a name like (IpitomyPBX)[[File:App options.png|none|frame]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Give your password a name and hit generate![[File:App password.png|none|frame]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Once you have generated that passcode it should be permanent unless the user deletes it from their Gmail account. Now you simply take that new password and imput it in the PBX and it should be good to go.[[File:Email config.png|none|frame]]&lt;br /&gt;
&lt;br /&gt;
If you are using the new PBX Version 7 software, and you are using Port 25, set SSL Support to No.&lt;br /&gt;
&lt;br /&gt;
[[File:Port 25.png|frameless]]&lt;br /&gt;
&amp;lt;/div&amp;gt;&lt;br /&gt;
&lt;br /&gt;
=== Using a Microsoft Office 365 Account ===&lt;br /&gt;
&lt;br /&gt;
The &amp;quot;Voicemail From Address&amp;quot; absolutely must be either a real email address or an alias configured.  You cannot use a nonexistent address even though your account login is real.  Use the following for the rest of the settings.&lt;br /&gt;
&lt;br /&gt;
*From Address: &amp;amp;lt;365 email address&amp;amp;gt;&lt;br /&gt;
*Voicemail Server: Ext&lt;br /&gt;
*Server Address: smtp.office365.com&lt;br /&gt;
*SSL Support: try NO first, and if it doesn't work change it to YES and try again.&lt;br /&gt;
*Server Port: 587&lt;br /&gt;
*Authentication Required: Yes&lt;br /&gt;
*User Name: &amp;amp;lt;365 login name/email&amp;amp;gt;&lt;br /&gt;
*Password: &amp;amp;lt;365 password&amp;amp;gt;&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=File:Port_25.png&amp;diff=5101</id>
		<title>File:Port 25.png</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=File:Port_25.png&amp;diff=5101"/>
		<updated>2024-05-16T15:03:29Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;jw&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPitomy_Cloud_Connect&amp;diff=5100</id>
		<title>IPitomy Cloud Connect</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPitomy_Cloud_Connect&amp;diff=5100"/>
		<updated>2024-05-15T14:27:33Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Sign In */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;&amp;lt;span style=&amp;quot;font-size: 12px;&amp;quot;&amp;gt;Note: This Guide will assist the user in Installing the &amp;lt;/span&amp;gt;[https://ipitomy.com/index.php/products/mobile-cloud-phone IPitomy Cloud Connect Softphone]&amp;amp;nbsp;&amp;lt;span style=&amp;quot;font-size: 12px;&amp;quot;&amp;gt;application on an &amp;lt;/span&amp;gt;[https://apps.apple.com/us/app/ipitomy-cloud-connect/id1534327260 iPhone]&amp;amp;nbsp;&amp;lt;span style=&amp;quot;font-size: 12px;&amp;quot;&amp;gt;and &amp;lt;/span&amp;gt;[https://play.google.com/store/apps/details?id=com.ipitomy.ipitomycloudconnect.android Android]&amp;amp;nbsp;&amp;lt;span style=&amp;quot;font-size: 12px;&amp;quot;&amp;gt;smartphone.&amp;lt;/span&amp;gt;&lt;br /&gt;
&lt;br /&gt;
== '''iPhone''' ==&lt;br /&gt;
&lt;br /&gt;
==== Search &amp;amp; Download ====&lt;br /&gt;
&lt;br /&gt;
Visit Apple AppStore and search &amp;quot;IPitomy&amp;quot; Press Get button to download application&lt;br /&gt;
&lt;br /&gt;
[[File:AppStore Screen270x480R.png]] [[File:App Screen270x480R.png]]&lt;br /&gt;
&lt;br /&gt;
==== &amp;lt;br/&amp;gt; Open Application ====&lt;br /&gt;
&lt;br /&gt;
Once application has finished installing press open button or tap on App icon to open&lt;br /&gt;
&lt;br /&gt;
[[File:OpenApp270x480.png]] [[File:OpenAppHS270x480.png]]&lt;br /&gt;
&lt;br /&gt;
==== Sign In ====&lt;br /&gt;
&lt;br /&gt;
Sign in by entering your username and password or scanning your QR Code (automatically logs into your account)&lt;br /&gt;
&lt;br /&gt;
[[File:SignIn.png]] [[File:OpenAppQR.png]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
If the APP turns off when you minimiz it, Background App Refresh has to be enabled.&lt;br /&gt;
&lt;br /&gt;
Also, Background App Refresh is automatically disabled if your phone is in low power mode.&lt;br /&gt;
&lt;br /&gt;
[[File:Background app refresh.png|frameless]]&lt;br /&gt;
== '''Android''' ==&lt;br /&gt;
&lt;br /&gt;
==== Search &amp;amp; Download ====&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Visit Google Play Store and search &amp;quot;IPitomy&amp;quot; Press Install button to download application&lt;br /&gt;
&lt;br /&gt;
[[File:1R.png]] [[File:2R.png]]&lt;br /&gt;
&lt;br /&gt;
==== &amp;lt;br/&amp;gt; Open Application ====&lt;br /&gt;
&lt;br /&gt;
Once application has finished installing press open button or tap on App icon to open&lt;br /&gt;
&lt;br /&gt;
[[File:3-1R.png]] [[File:3R.png]]&lt;br /&gt;
&lt;br /&gt;
==== Sign In ====&lt;br /&gt;
&lt;br /&gt;
Sign in by entering username and password or scanning QR Code (automatically logs into your account)&lt;br /&gt;
&lt;br /&gt;
[[File:4R.png]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
PROXY SERVER IP's:&lt;br /&gt;
&lt;br /&gt;
165.227.0.0/16&lt;br /&gt;
&lt;br /&gt;
167.99.0.0/16&lt;br /&gt;
&lt;br /&gt;
159.65.0.0/16&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IPitomy_Cloud_Connect&amp;diff=5099</id>
		<title>IPitomy Cloud Connect</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IPitomy_Cloud_Connect&amp;diff=5099"/>
		<updated>2024-05-15T14:24:46Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: /* Sign In */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;&amp;lt;span style=&amp;quot;font-size: 12px;&amp;quot;&amp;gt;Note: This Guide will assist the user in Installing the &amp;lt;/span&amp;gt;[https://ipitomy.com/index.php/products/mobile-cloud-phone IPitomy Cloud Connect Softphone]&amp;amp;nbsp;&amp;lt;span style=&amp;quot;font-size: 12px;&amp;quot;&amp;gt;application on an &amp;lt;/span&amp;gt;[https://apps.apple.com/us/app/ipitomy-cloud-connect/id1534327260 iPhone]&amp;amp;nbsp;&amp;lt;span style=&amp;quot;font-size: 12px;&amp;quot;&amp;gt;and &amp;lt;/span&amp;gt;[https://play.google.com/store/apps/details?id=com.ipitomy.ipitomycloudconnect.android Android]&amp;amp;nbsp;&amp;lt;span style=&amp;quot;font-size: 12px;&amp;quot;&amp;gt;smartphone.&amp;lt;/span&amp;gt;&lt;br /&gt;
&lt;br /&gt;
== '''iPhone''' ==&lt;br /&gt;
&lt;br /&gt;
==== Search &amp;amp; Download ====&lt;br /&gt;
&lt;br /&gt;
Visit Apple AppStore and search &amp;quot;IPitomy&amp;quot; Press Get button to download application&lt;br /&gt;
&lt;br /&gt;
[[File:AppStore Screen270x480R.png]] [[File:App Screen270x480R.png]]&lt;br /&gt;
&lt;br /&gt;
==== &amp;lt;br/&amp;gt; Open Application ====&lt;br /&gt;
&lt;br /&gt;
Once application has finished installing press open button or tap on App icon to open&lt;br /&gt;
&lt;br /&gt;
[[File:OpenApp270x480.png]] [[File:OpenAppHS270x480.png]]&lt;br /&gt;
&lt;br /&gt;
==== Sign In ====&lt;br /&gt;
&lt;br /&gt;
Sign in by entering your username and password or scanning your QR Code (automatically logs into your account)&lt;br /&gt;
&lt;br /&gt;
[[File:SignIn.png]] [[File:OpenAppQR.png]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
If the APP turns off when you minimiz it, Background App Refresh has to be enabled.&lt;br /&gt;
&lt;br /&gt;
[[File:Background app refresh.png|frameless]]&lt;br /&gt;
== '''Android''' ==&lt;br /&gt;
&lt;br /&gt;
==== Search &amp;amp; Download ====&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Visit Google Play Store and search &amp;quot;IPitomy&amp;quot; Press Install button to download application&lt;br /&gt;
&lt;br /&gt;
[[File:1R.png]] [[File:2R.png]]&lt;br /&gt;
&lt;br /&gt;
==== &amp;lt;br/&amp;gt; Open Application ====&lt;br /&gt;
&lt;br /&gt;
Once application has finished installing press open button or tap on App icon to open&lt;br /&gt;
&lt;br /&gt;
[[File:3-1R.png]] [[File:3R.png]]&lt;br /&gt;
&lt;br /&gt;
==== Sign In ====&lt;br /&gt;
&lt;br /&gt;
Sign in by entering username and password or scanning QR Code (automatically logs into your account)&lt;br /&gt;
&lt;br /&gt;
[[File:4R.png]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
PROXY SERVER IP's:&lt;br /&gt;
&lt;br /&gt;
165.227.0.0/16&lt;br /&gt;
&lt;br /&gt;
167.99.0.0/16&lt;br /&gt;
&lt;br /&gt;
159.65.0.0/16&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=File:Background_app_refresh.png&amp;diff=5098</id>
		<title>File:Background app refresh.png</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=File:Background_app_refresh.png&amp;diff=5098"/>
		<updated>2024-05-15T14:24:13Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;jw&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IP_PBX_Manual_PBXSetup_SIP&amp;diff=5092</id>
		<title>IP PBX Manual PBXSetup SIP</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IP_PBX_Manual_PBXSetup_SIP&amp;diff=5092"/>
		<updated>2024-03-21T17:56:21Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;__NOTITLE__&lt;br /&gt;
{{IP_PBX_Manual|sortkey=SIP}}&lt;br /&gt;
== SIP Setup&amp;lt;br/&amp;gt; ==&lt;br /&gt;
&lt;br /&gt;
'''This page defines global SIP configuration parameters.'''&lt;br /&gt;
&lt;br /&gt;
=== [[File:Sip page.png|alt=|File:PBX Setup-SIP.jpg]]  ===&lt;br /&gt;
&lt;br /&gt;
=== &amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
=== SIP Networking Settings Section&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Sections/Fields'''&amp;lt;/center&amp;gt;&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;&lt;br /&gt;
'''Description'''&lt;br /&gt;
&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Network Address Included'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | The PBX automatically includes its own localnet if the setting below is set to yes.&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Local Network &amp;amp; Subnet Masks'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | This is where you can add additional IP address to the local network and associated subnet masks.&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Delete Selected'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allows you to delete the item selected from list of networks.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Add Local Network'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allows you to add local network information. This information will appear in the list of networks.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Include LAN network as Localnet'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | When set to Yes, the PBX will automatically include its own localnet.&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Remote Clients Access PBX by'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Defines how Remote Clients (remote phones or SIP providers) access the PBX.&amp;amp;nbsp; Set to &amp;quot;'''No Remote'''&amp;quot; if neither is being used.&amp;amp;nbsp; Set to &amp;quot;'''IP address'''&amp;quot; if the site has a static IP or set to &amp;quot;'''Hostname'''&amp;quot; if the site uses a dynamic IP. &amp;amp;nbsp; For the last option to work, they will need a Dyndns.com domain, and enter that as the External Host. &amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''External IP'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Used if &amp;quot;'''Remote Clients Access PBX by'''&amp;quot; is set to IP Address. Click the Get IP button to populate this field with the public IP of the gateway the PBX uses to access the Internet&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''External Host'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Used if &amp;quot;'''Remote Clients Access PBX by'''&amp;quot; is set to Hostname.&amp;amp;nbsp; Enter the dyndns.com domain that resolves to the current dynamic IP of the system here.&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''External Host Refresh'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Used if &amp;quot;'''Remote Clients Access PBX by'''&amp;quot; is set to Hostname.&amp;amp;nbsp; This is the interval which the PBX will check for an up to date IP.&amp;amp;nbsp; Set this to a value within the interval set at dyndns, typically between 300 and 3000.&amp;lt;br/&amp;gt;&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
==== Add SIP Networking Settings ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup-&amp;gt;SIP '''page, locate the '''SIP Networking Settings '''section.&lt;br /&gt;
#Enter the IP Address and Subnet Mask for the network the PBX is being installed on.&lt;br /&gt;
#Click the Add button.&lt;br /&gt;
#Click the [[File:Savechanges.png]] button.&lt;br /&gt;
#Click on the '''Apply Changes''' link at the top of the page to save the information and commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
==== Delete SIP Networking Settings ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup-&amp;gt;SIP '''page, locate the '''SIP Networking Settings '''section.&lt;br /&gt;
#Highlight the listing you wish to delete. You can use Shift/Ctrl click functionality to select multiple listings.&lt;br /&gt;
#Click the Delete Selected button.&lt;br /&gt;
#Click the [[File:Savechanges.png]] button.&lt;br /&gt;
#Click on the '''Apply Changes''' link at the top of the page to save the information and commit the changes to the database&lt;br /&gt;
&lt;br /&gt;
=== SIP Advanced Settings Section ===&lt;br /&gt;
&lt;br /&gt;
Advanced SIP settings define in more detail the management of network traffic. These settings are automatically provisioned when the system registers with the router. In most business implementations it is not necessary to make changes to these defaulted settings&lt;br /&gt;
[[File:SIPSettings1.png|alt=|center|frameless|748x748px]]&lt;br /&gt;
[[File:SIPSettings2.png|alt=|center|frameless|844x844px]]&lt;br /&gt;
[[File:SIPSettings3.png|alt=|center|frameless|545x545px]]&lt;br /&gt;
&lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #000000;  border-bottom:0.0069in solid #000000;  border-left:0.0069in solid #000000;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #000000;  border-bottom:0.0069in solid #000000;  border-left:none;  border-right:0.0069in solid #000000;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''IMPORTANT:''' '''The default settings for the SIP configuration should not require any changes. If it is necessary for you to do so to meet your customer’s business requirements, we recommend that you contact IPitomy’s Technical Support for assistance.'''&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
{| class=&amp;quot;wikitable&amp;quot; style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''&amp;lt;big&amp;gt;Sections/Fields&amp;lt;/big&amp;gt;'''&amp;lt;/center&amp;gt;&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''&amp;lt;big&amp;gt;Description/Default Parameters&amp;lt;/big&amp;gt;'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Call Context&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''INCOMING'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow Guest Calls&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Host/Domain Name&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | UDP Port&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''5060'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Bind Address&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''0.0.0.0'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Enable DNS SRV Lookup&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''NO'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Domains&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow External Invites&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Auto Domain&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Enable Pedantic Checking&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | SIP TOS&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default is '''CS3'''. To configure QOS on your LAN, set your managed switches to prioritize packets flagged with CS3&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | RTP TOS&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default is '''CS3'''. To configure QOS on your LAN, set your managed switches to prioritize packets flagged with CS3&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Video TOS&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default is '''CS3'''. To configure QOS on your LAN, set your managed switches to prioritize packets flagged with CS3&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Max Length of Registration&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''7200'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default Length of Registration&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''3600'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Notify Mime Type&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Time Between Mailbox Checks&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Voicemail Extension&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | SIP Video Support&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Record History of Default&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | First disallow all Codecs&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''ALL'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow Codecs&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''G.711 ulaw, G.711 alaw, GSM'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default Music on Hold&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | This will display whatever playlist is set to default on the PBX Setup=&amp;gt;Music On Hold page&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Relax DTMF Handling&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | RTP Keep-Alive&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Sends RTP packet when none received on active call for X seconds, 0 for disabled, which is the default.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | RTP Timeout&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default is '''BLANK'''. Set to a value, in seconds, if you wish the PBX to end a call when no RTP traffic is detected for that long. Typically used in regards to lines that are not disconnecting correctly.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | RTP Timeout on Hold&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Trust Remote Party ID&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Send Remote Party ID&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Progress in Band&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | User Agent&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow Redirect to Non-local SIP Address&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | User = Phone&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | DTMF Mode&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''AUTO'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Compact SIP Headers&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | SIP Debug&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Subscriber Context&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Notify Ringing&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Qualify&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: 8000&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Generate Manager Events&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | NAT&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Insecure&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''VERY'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Can Reinvite&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Cache Realtime Friends&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Real Time Update&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Auto-Expire Friends&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Ignore Registration Expiration&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow External Domains&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
|RTP Start&lt;br /&gt;
|Default: 10000 (Defines the starting port range for RTP Traffic. This should only be changed if your trunk provider uses different ports to establish RTP [audio] feed.)&lt;br /&gt;
|-&lt;br /&gt;
|RTP End&lt;br /&gt;
|Default: 20000 (Defines the ending port range for RTP Traffic. This should only be changed if your trunk provider uses different ports to establish RTP [audio] feed.)&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
==== Edit Advanced SIP Networking Settings ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup-&amp;gt;SIP '''page, click on the '''Advanced '''link.&lt;br /&gt;
#The '''Advanced SIP Networking Settings''' page is displayed.&lt;br /&gt;
#Set the '''SIP Network''' parameters base on your business requirements or what is recommended by IPitomy.&lt;br /&gt;
The default settings should not require any changes. If it is necessary for you to do so to meet your customer’s business requirements, we recommend that you contact IPitomy’s Technical Support for assistance..&lt;br /&gt;
&lt;br /&gt;
#Click the [[File:Savechanges.png]] button.&lt;br /&gt;
#Click on the '''Apply Changes''' link at the top of the page to save the information and commit the changes to the database.&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IP_PBX_Manual_PBXSetup_SIP&amp;diff=5091</id>
		<title>IP PBX Manual PBXSetup SIP</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IP_PBX_Manual_PBXSetup_SIP&amp;diff=5091"/>
		<updated>2024-03-21T17:53:57Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;__NOTITLE__&lt;br /&gt;
{{IP_PBX_Manual|sortkey=SIP}}&lt;br /&gt;
== SIP Setup&amp;lt;br/&amp;gt; ==&lt;br /&gt;
&lt;br /&gt;
'''This page defines global SIP configuration parameters.'''&lt;br /&gt;
&lt;br /&gt;
=== [[File:Sip page.png|alt=|File:PBX Setup-SIP.jpg]]  ===&lt;br /&gt;
&lt;br /&gt;
=== &amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
=== SIP Networking Settings Section&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Sections/Fields'''&amp;lt;/center&amp;gt;&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;&lt;br /&gt;
'''Description'''&lt;br /&gt;
&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Network Address Included'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | The PBX automatically includes its own localnet if the setting below is set to yes.&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Local Network &amp;amp; Subnet Masks'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | This is where you can add additional IP address to the local network and associated subnet masks.&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Delete Selected'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allows you to delete the item selected from list of networks.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Add Local Network'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allows you to add local network information. This information will appear in the list of networks.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Include LAN network as Localnet'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | When set to Yes, the PBX will automatically include its own localnet.&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Remote Clients Access PBX by'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Defines how Remote Clients (remote phones or SIP providers) access the PBX.&amp;amp;nbsp; Set to &amp;quot;'''No Remote'''&amp;quot; if neither is being used.&amp;amp;nbsp; Set to &amp;quot;'''IP address'''&amp;quot; if the site has a static IP or set to &amp;quot;'''Hostname'''&amp;quot; if the site uses a dynamic IP. &amp;amp;nbsp; For the last option to work, they will need a Dyndns.com domain, and enter that as the External Host. &amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''External IP'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Used if &amp;quot;'''Remote Clients Access PBX by'''&amp;quot; is set to IP Address. Click the Get IP button to populate this field with the public IP of the gateway the PBX uses to access the Internet&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''External Host'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Used if &amp;quot;'''Remote Clients Access PBX by'''&amp;quot; is set to Hostname.&amp;amp;nbsp; Enter the dyndns.com domain that resolves to the current dynamic IP of the system here.&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''External Host Refresh'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Used if &amp;quot;'''Remote Clients Access PBX by'''&amp;quot; is set to Hostname.&amp;amp;nbsp; This is the interval which the PBX will check for an up to date IP.&amp;amp;nbsp; Set this to a value within the interval set at dyndns, typically between 300 and 3000.&amp;lt;br/&amp;gt;&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
==== Add SIP Networking Settings ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup-&amp;gt;SIP '''page, locate the '''SIP Networking Settings '''section.&lt;br /&gt;
#Enter the IP Address and Subnet Mask for the network the PBX is being installed on.&lt;br /&gt;
#Click the Add button.&lt;br /&gt;
#Click the [[File:Savechanges.png]] button.&lt;br /&gt;
#Click on the '''Apply Changes''' link at the top of the page to save the information and commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
==== Delete SIP Networking Settings ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup-&amp;gt;SIP '''page, locate the '''SIP Networking Settings '''section.&lt;br /&gt;
#Highlight the listing you wish to delete. You can use Shift/Ctrl click functionality to select multiple listings.&lt;br /&gt;
#Click the Delete Selected button.&lt;br /&gt;
#Click the [[File:Savechanges.png]] button.&lt;br /&gt;
#Click on the '''Apply Changes''' link at the top of the page to save the information and commit the changes to the database&lt;br /&gt;
&lt;br /&gt;
=== SIP Advanced Settings Section ===&lt;br /&gt;
&lt;br /&gt;
Advanced SIP settings define in more detail the management of network traffic. These settings are automatically provisioned when the system registers with the router. In most business implementations it is not necessary to make changes to these defaulted settings&lt;br /&gt;
&lt;br /&gt;
[[File:SIPSettings1.png|frameless|748x748px]]&lt;br /&gt;
&lt;br /&gt;
[[File:SIPSettings2.png|frameless|844x844px]]&lt;br /&gt;
&lt;br /&gt;
[[File:SIPSettings3.png|frameless|545x545px]]&lt;br /&gt;
&lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #000000;  border-bottom:0.0069in solid #000000;  border-left:0.0069in solid #000000;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #000000;  border-bottom:0.0069in solid #000000;  border-left:none;  border-right:0.0069in solid #000000;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''IMPORTANT:''' '''The default settings for the SIP configuration should not require any changes. If it is necessary for you to do so to meet your customer’s business requirements, we recommend that you contact IPitomy’s Technical Support for assistance.'''&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
{| class=&amp;quot;wikitable&amp;quot; style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''&amp;lt;big&amp;gt;Sections/Fields&amp;lt;/big&amp;gt;'''&amp;lt;/center&amp;gt;&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''&amp;lt;big&amp;gt;Description/Default Parameters&amp;lt;/big&amp;gt;'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Call Context&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''INCOMING'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow Guest Calls&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Host/Domain Name&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | UDP Port&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''5060'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Bind Address&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''0.0.0.0'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Enable DNS SRV Lookup&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''NO'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Domains&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow External Invites&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Auto Domain&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Enable Pedantic Checking&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | SIP TOS&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default is '''CS3'''. To configure QOS on your LAN, set your managed switches to prioritize packets flagged with CS3&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | RTP TOS&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default is '''CS3'''. To configure QOS on your LAN, set your managed switches to prioritize packets flagged with CS3&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Video TOS&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default is '''CS3'''. To configure QOS on your LAN, set your managed switches to prioritize packets flagged with CS3&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Max Length of Registration&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''7200'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default Length of Registration&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''3600'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Notify Mime Type&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Time Between Mailbox Checks&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Voicemail Extension&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | SIP Video Support&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Record History of Default&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | First disallow all Codecs&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''ALL'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow Codecs&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''G.711 ulaw, G.711 alaw, GSM'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default Music on Hold&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | This will display whatever playlist is set to default on the PBX Setup=&amp;gt;Music On Hold page&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Relax DTMF Handling&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | RTP Keep-Alive&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Sends RTP packet when none received on active call for X seconds, 0 for disabled, which is the default.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | RTP Timeout&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default is '''BLANK'''. Set to a value, in seconds, if you wish the PBX to end a call when no RTP traffic is detected for that long. Typically used in regards to lines that are not disconnecting correctly.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | RTP Timeout on Hold&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Trust Remote Party ID&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Send Remote Party ID&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Progress in Band&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | User Agent&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow Redirect to Non-local SIP Address&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | User = Phone&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | DTMF Mode&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''AUTO'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Compact SIP Headers&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | SIP Debug&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Subscriber Context&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Notify Ringing&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Qualify&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: 8000&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Generate Manager Events&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | NAT&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Insecure&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''VERY'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Can Reinvite&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Cache Realtime Friends&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Real Time Update&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Auto-Expire Friends&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Ignore Registration Expiration&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow External Domains&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
|RTP Start&lt;br /&gt;
|Default: 10000 (Defines the starting port range for RTP Traffic. This should only be changed if your trunk provider uses different ports to establish RTP [audio] feed.)&lt;br /&gt;
|-&lt;br /&gt;
|RTP End&lt;br /&gt;
|Default: 20000 (Defines the ending port range for RTP Traffic. This should only be changed if your trunk provider uses different ports to establish RTP [audio] feed.)&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
==== Edit Advanced SIP Networking Settings ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup-&amp;gt;SIP '''page, click on the '''Advanced '''link.&lt;br /&gt;
#The '''Advanced SIP Networking Settings''' page is displayed.&lt;br /&gt;
#Set the '''SIP Network''' parameters base on your business requirements or what is recommended by IPitomy.&lt;br /&gt;
The default settings should not require any changes. If it is necessary for you to do so to meet your customer’s business requirements, we recommend that you contact IPitomy’s Technical Support for assistance..&lt;br /&gt;
&lt;br /&gt;
#Click the [[File:Savechanges.png]] button.&lt;br /&gt;
#Click on the '''Apply Changes''' link at the top of the page to save the information and commit the changes to the database.&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=IP_PBX_Manual_PBXSetup_SIP&amp;diff=5090</id>
		<title>IP PBX Manual PBXSetup SIP</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=IP_PBX_Manual_PBXSetup_SIP&amp;diff=5090"/>
		<updated>2024-03-21T17:11:55Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;__NOTITLE__&lt;br /&gt;
{{IP_PBX_Manual|sortkey=SIP}}&lt;br /&gt;
== SIP Setup&amp;lt;br/&amp;gt; ==&lt;br /&gt;
&lt;br /&gt;
'''This page defines global SIP configuration parameters.'''&lt;br /&gt;
&lt;br /&gt;
=== [[File:Sip page.png|alt=|File:PBX Setup-SIP.jpg]]  ===&lt;br /&gt;
&lt;br /&gt;
=== &amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
=== SIP Networking Settings Section&amp;lt;br/&amp;gt; ===&lt;br /&gt;
&lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Sections/Fields'''&amp;lt;/center&amp;gt;&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;&lt;br /&gt;
'''Description'''&lt;br /&gt;
&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Network Address Included'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | The PBX automatically includes its own localnet if the setting below is set to yes.&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Local Network &amp;amp; Subnet Masks'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | This is where you can add additional IP address to the local network and associated subnet masks.&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Delete Selected'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allows you to delete the item selected from list of networks.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Add Local Network'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allows you to add local network information. This information will appear in the list of networks.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Include LAN network as Localnet'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | When set to Yes, the PBX will automatically include its own localnet.&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''Remote Clients Access PBX by'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Defines how Remote Clients (remote phones or SIP providers) access the PBX.&amp;amp;nbsp; Set to &amp;quot;'''No Remote'''&amp;quot; if neither is being used.&amp;amp;nbsp; Set to &amp;quot;'''IP address'''&amp;quot; if the site has a static IP or set to &amp;quot;'''Hostname'''&amp;quot; if the site uses a dynamic IP. &amp;amp;nbsp; For the last option to work, they will need a Dyndns.com domain, and enter that as the External Host. &amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''External IP'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Used if &amp;quot;'''Remote Clients Access PBX by'''&amp;quot; is set to IP Address. Click the Get IP button to populate this field with the public IP of the gateway the PBX uses to access the Internet&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''External Host'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Used if &amp;quot;'''Remote Clients Access PBX by'''&amp;quot; is set to Hostname.&amp;amp;nbsp; Enter the dyndns.com domain that resolves to the current dynamic IP of the system here.&amp;lt;br/&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-width: 0.0069in medium 0.0069in 0.0069in;  border-style: solid none solid solid;  border-color: rgb(0, 0, 255) -moz-use-text-color rgb(0, 0, 255) rgb(0, 0, 255);  padding: 0in 0.075in;  text-align: center&amp;quot; | '''External Host Refresh'''&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Used if &amp;quot;'''Remote Clients Access PBX by'''&amp;quot; is set to Hostname.&amp;amp;nbsp; This is the interval which the PBX will check for an up to date IP.&amp;amp;nbsp; Set this to a value within the interval set at dyndns, typically between 300 and 3000.&amp;lt;br/&amp;gt;&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
==== Add SIP Networking Settings ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup-&amp;gt;SIP '''page, locate the '''SIP Networking Settings '''section.&lt;br /&gt;
#Enter the IP Address and Subnet Mask for the network the PBX is being installed on.&lt;br /&gt;
#Click the Add button.&lt;br /&gt;
#Click the [[File:Savechanges.png]] button.&lt;br /&gt;
#Click on the '''Apply Changes''' link at the top of the page to save the information and commit the changes to the database.&lt;br /&gt;
&lt;br /&gt;
==== Delete SIP Networking Settings ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup-&amp;gt;SIP '''page, locate the '''SIP Networking Settings '''section.&lt;br /&gt;
#Highlight the listing you wish to delete. You can use Shift/Ctrl click functionality to select multiple listings.&lt;br /&gt;
#Click the Delete Selected button.&lt;br /&gt;
#Click the [[File:Savechanges.png]] button.&lt;br /&gt;
#Click on the '''Apply Changes''' link at the top of the page to save the information and commit the changes to the database&lt;br /&gt;
&lt;br /&gt;
=== SIP Advanced Settings Section ===&lt;br /&gt;
&lt;br /&gt;
Advanced SIP settings define in more detail the management of network traffic. These settings are automatically provisioned when the system registers with the router. In most business implementations it is not necessary to make changes to these defaulted settings&lt;br /&gt;
&lt;br /&gt;
[[File:SIPSettings1.png|frameless|748x748px]]&lt;br /&gt;
&lt;br /&gt;
[[File:SIPSettings2.png|frameless|844x844px]]&lt;br /&gt;
&lt;br /&gt;
[[File:SIPSettings3.png|frameless|545x545px]]&lt;br /&gt;
&lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #000000;  border-bottom:0.0069in solid #000000;  border-left:0.0069in solid #000000;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #000000;  border-bottom:0.0069in solid #000000;  border-left:none;  border-right:0.0069in solid #000000;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | '''IMPORTANT:''' '''The default settings for the SIP configuration should not require any changes. If it is necessary for you to do so to meet your customer’s business requirements, we recommend that you contact IPitomy’s Technical Support for assistance.'''&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
{| style=&amp;quot;border-spacing:0&amp;quot;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Sections/Fields'''&amp;lt;/center&amp;gt;&lt;br /&gt;
| style=&amp;quot;background-color:#b8cce4;  border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | &amp;lt;center&amp;gt;'''Description/Default Parameters'''&amp;lt;/center&amp;gt;&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Call Context&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''INCOMING'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow Guest Calls&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Host/Domain Name&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | UDP Port&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''5060'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Bind Address&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''0.0.0.0'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Enable DNS SRV Lookup&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''NO'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Domains&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow External Invites&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Auto Domain&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Enable Pedantic Checking&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | SIP TOS&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default is '''CS3'''. To configure QOS on your LAN, set your managed switches to prioritize packets flagged with CS3&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | RTP TOS&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default is '''CS3'''. To configure QOS on your LAN, set your managed switches to prioritize packets flagged with CS3&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Video TOS&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default is '''CS3'''. To configure QOS on your LAN, set your managed switches to prioritize packets flagged with CS3&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Max Length of Registration&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''7200'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default Length of Registration&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''3600'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Notify Mime Type&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Time Between Mailbox Checks&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Voicemail Extension&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | SIP Video Support&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Record History of Default&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | First disallow all Codecs&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''ALL'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow Codecs&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''G.711 ulaw, G.711 alaw, GSM'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default Music on Hold&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | This will display whatever playlist is set to default on the PBX Setup=&amp;gt;Music On Hold page&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Relax DTMF Handling&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | RTP Keep-Alive&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Sends RTP packet when none received on active call for X seconds, 0 for disabled, which is the default.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | RTP Timeout&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default is '''BLANK'''. Set to a value, in seconds, if you wish the PBX to end a call when no RTP traffic is detected for that long. Typically used in regards to lines that are not disconnecting correctly.&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | RTP Timeout on Hold&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Trust Remote Party ID&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Send Remote Party ID&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Progress in Band&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | User Agent&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow Redirect to Non-local SIP Address&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | User = Phone&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | DTMF Mode&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''AUTO'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Compact SIP Headers&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | SIP Debug&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Subscriber Context&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''BLANK'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Notify Ringing&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Qualify&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: 8000&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Generate Manager Events&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | NAT&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Insecure&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''VERY'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Can Reinvite&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Cache Realtime Friends&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Real Time Update&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Auto-Expire Friends&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Ignore Registration Expiration&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''N/A'''&lt;br /&gt;
|-&lt;br /&gt;
| style=&amp;quot;border-top:0.0069in solid #0000ff;  border-bottom:0.0069in solid #0000ff;  border-left:0.0069in solid #0000ff;  border-right:none;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Allow External Domains&amp;lt;br/&amp;gt;&lt;br /&gt;
| style=&amp;quot;border:0.0069in solid #0000ff;  padding-top:0in;  padding-bottom:0in;  padding-left:0.075in;  padding-right:0.075in&amp;quot; | Default: '''YES'''&lt;br /&gt;
|-&lt;br /&gt;
|RTP Start&lt;br /&gt;
|Default: 10000 (Defines starting port range for RTP Traffic. This should only be changed if your trunk provider uses different ports to establish RTP (audio) feed.&lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
==== Edit Advanced SIP Networking Settings ====&lt;br /&gt;
&lt;br /&gt;
'''STEPS:'''&lt;br /&gt;
&lt;br /&gt;
#From the '''PBX Setup-&amp;gt;SIP '''page, click on the '''Advanced '''link.&lt;br /&gt;
#The '''Advanced SIP Networking Settings''' page is displayed.&lt;br /&gt;
#Set the '''SIP Network''' parameters base on your business requirements or what is recommended by IPitomy.&lt;br /&gt;
The default settings should not require any changes. If it is necessary for you to do so to meet your customer’s business requirements, we recommend that you contact IPitomy’s Technical Support for assistance..&lt;br /&gt;
&lt;br /&gt;
#Click the [[File:Savechanges.png]] button.&lt;br /&gt;
#Click on the '''Apply Changes''' link at the top of the page to save the information and commit the changes to the database.&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=File:SIPSettings3.png&amp;diff=5089</id>
		<title>File:SIPSettings3.png</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=File:SIPSettings3.png&amp;diff=5089"/>
		<updated>2024-03-21T16:42:24Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;SIPSettings3&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
	<entry>
		<id>http://wiki.ipitomy.com/index.php?title=File:SIPSettings2.png&amp;diff=5088</id>
		<title>File:SIPSettings2.png</title>
		<link rel="alternate" type="text/html" href="http://wiki.ipitomy.com/index.php?title=File:SIPSettings2.png&amp;diff=5088"/>
		<updated>2024-03-21T16:39:05Z</updated>

		<summary type="html">&lt;p&gt;Mike Lunn: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;SIPSettings2&lt;/div&gt;</summary>
		<author><name>Mike Lunn</name></author>
	</entry>
</feed>