IPitomy Wiki - User contributions [en]

HD Phone FAQ

2015-12-22T18:07:20Z

John Wolfe:

__NOTITLE__

== HD Phone FAQ ==

=== I have an IP330 on a gigabit switch and I am periodically having issues with connectivity from a PC connected through the phone. Is there anything I can do to resolve the issue? ===

On some older Network cards we have had reports of trouble when they go to sleep. This is apparently a common problem with an older RealTek network cards.

Here are the things that we have seen work in the field:

*Replace the cable with a gigabit cable. There are greater requirements upon the quality of the cable when supporting gigabit.
*In the hardware properties of the network card in windows, disable sleep. (In Win7 this is under Power Management -> Allow computer to turn off this device to save power).
*Set the speed to a fixed value in the PC, try 1000 or 10/100

=== How can I download logs from the HD phone? ===

Sometimes IPitomy will require you to get the logs from your phone so we can investigate an issue more deeply. Steps to accomplish this are:

#Log into the IP address of the phone (username: root/password: root)
#Navigate to Phone Maintenance=>HTTP Upgrade
#Click the Download button next to All Config File:

NOTE:It may take a few minutes to download all of the files, be patient.

=== How do I configure my my HD Phones to use VLANs? ===

If you need to use VLAN to control what IP addresses the phones pull from DHCP you can modify the Phone Configuration Template in the PBX. VLANs logically isolate network traffic, they do not necessarily physically isolate it. It is typically going to be used because there are more devices on the network then there are IP Addresses in a single subnet. Or in order to implement security.

#Navigate to PBX Setup=>Phone Global
#Click Edit Current Template
#Find the following section in the template pertains to VLAN, this is how it appears by default

<code>
<VLANs>
<Vlan LocalEnableVlan="0" PCEnableVlan="0" LocalVID="0" PCVID="0" LocalPriority="0" PCPriority="0" EnableVlan="0"/>
</VLANs>
</code>

{| class="wikitable"
|-
! Variable
! Function
! Settings
|-
| LocalEnableVlan
| Controls if the LAN port is using vLAN tagging
| 0 = Disabled, 1 = Enabled
|-
| PCEnableVlan
| Controls if the PC port is using vLAN tagging
| 0 = Disabled, 1 = Enabled
|-
| LocalVID
| Defines what vLAN tag the LAN port is using
| Set to the vLAN ID
|-
| PCVID
| Defines what vLAN tag the PC port is using
| Set to the vLAN ID
|-
| LocalPriority
| Not Used
| Leave at 0
|-
| PC Priority
| Not Used
| Leave at 0
|-
| EnableVlan
| Global setting to toggle vLAN on PC and LAN port
| 0 = Disabled, 1 = Enabled
|}

*In the following example the LAN port will use VLAN 5 and the PC port will use VLAN 8, allowing the Phone and PC to get different IP addresses from the DHCP server:

<VLANs>
<Vlan LocalEnableVlan="1" PCEnableVlan="1" LocalVID="5" PCVID="8" LocalPriority="0" PCPriority="0" EnableVlan="1"/>
</VLANs>

*When done modifying the template Save, Apply Changes, and now any phones that are configured to the PBX will have their configuration file modified to match the template changes.

NOTE: If the phones are default, you may have to statically set them to use the correct VLAN so they can communicate with the PBX to get their configuration.

=== How do I Enabled LLDP Globally for the Phones ===

In the Global Template (PBX Setup=>Phone Global) you can search for LLDP and will see <code>LLDP="0"</code>, this designates that LLDP is disabled. In order to enable you will need to change the 0 to a 1. You will also need to add the following <code> LLDPInterval="60" </code>.

When all is set the entry in the Global Template will look like this:
<code>LLDP="1" LLDPInterval="60"</code>

Click this link for general instructions on setting up [[LLDP]] in the switch.

=== How can I set my Phone to use a static IP address? ===

IPitomy supports configuring phones using DHCP and this is the best option for almost every network. If you would like to statically set IP addresses, we recommend that you use DHCP reservations to do this. In the vast majority of routers and DHCP servers there is an ability to assign a static IP address to a MAC address. While this process of assigning may be tedious, it is no less so than going to each individual phone and manually configuring the IP address (Static IP). In the end it eliminates any difficult modifications that you will need to make to configuration files and is just a way easier method to use for accomplishing the same task.

If having this information, the phone still must be manually set to static IP addresses then you will need to understand the following information. You have several options for performing this configuration:

==== Configure then Disable Configuration Updates ====

#Automatically Configure the phones using DHCP
#Go to phone's web page (root/root to log in) Navigate to Phone Maintenance->Auto Provision and uncheck the box that says Auto Download Config File.
#set the IP address to the desired static address
#The phone should keep it's original configuration and any changes made to phone configuration will now have to be done through the phone web interface. The exception to this is that you can change the phone keys in this configuration, just do NOT press Save and Configure. Press Save then Apply changes. The phone will update the keys at night when it downloads it's key configuration.

==== Change Configuration File Template ====

'''Please read the section of the Documentation [[http://wiki.ipitomy.com/index.php/HD_Phones#A_Note_on_Changing_the_Global_Template Note on Changing the Global Template]] to understand the implications of using this method.'''

The second option is more difficult to initially configure but is easier because you can still change phone settings. You should only do this if you are familiar with XML as you can break the configuration files by making changes here that lead to invalid XML.

#Go to PBXSetup->Phone Global
#Under "HD Phone Configuration Template" click "Reset Template to Default"
#Now click Edit Template. Under Networks - delete the entire Node called "network"
#Now phones should keep whatever you manually set them to.
#Note with this method that when you upgrade the PBX software you should be aware that we may make changes to the default template and you may have to reset the template and remake the changes in order to support additional functionality that is added in future releases.

The final option here is to just manually configure the phone. This works but if you call tech support with an issue they will probably have you default and reconfigure the phone through the PBX as we change a number of options when we provision the phones to the PBX.

=== Can I use HD Phones with my IP1000? ===

''YES.  HD phones will function normally as most other SIP phones with your IP1000. ''

In order to configure the phone you must use [[HD Phones#Manual|Manual Phone Configuration]].

=== My BLF won’t light on my phone? ===

If you have a phone where a BLF will not come on when it is supposed to, there are several things you can do.  The most important thing is to understand is how a BLF is turned on or off.  When a phone goes off hook and connects with a call, it sends out a short status packet to the PBX.  The packet simply says that the status has changed from available to busy.  The PBX then updates all of the phones that have a subscription to the BLF with a status packet indicating that the status has changed to busy.

The packet ether got there or it didn’t.  If it didn’t, the status will not update.  The status should update automatically every 30 minutes or so.  If the packet was missed, everything should be back to normal the next time the phone re-subscribes (about every 30 minutes automatically).  

What can go wrong?

The packet might have been lost.  What do I do? Why would the packet get lost in the first place?  Is it magic?  Is it being controlled by “The Force”?  Probably not.  The packet may have not arrived due to many factors.  Most likely the flow of normal packets was disrupted.  In the world of SIP, signaling packets have a very short life.  They become stale if they are not delivered on time.  The SIP packets are delivered using UDP.  This protocol does not ask for a positive acknowledgement when a packet is received.  It is assumed that it has been received.  There are several things that can cause a packet to be delayed on a LAN.  All these things have the same symptom.  

Here is a short troubleshooting guide:

#Perhaps you have recently restarted all services or rebooted your PBX.  You phones lose their BLF subscriptions when this occurs.  The phones will resubscribe automatically in time.  If you wish to have the BLFs work immediately then you will need to reboot the effected phones.
#The packet might not be the culprit; maybe your lamp is broken.  Test this by going to the button on the phone and select diagnostics.  Select the option lights.  Did the light come on?  If it did, it’s not the light.
#To narrow this down a little further, it is best to remove all doubt.  Restore the phone to factory defaults and reprogram it.  Test it and see if the problem happens again.  If it does, go to the next step.
#Troubleshoot the network.  See this guide for assistance [[Troubleshooting: Network|Network Troubleshooting]] 

There is no such thing as black magic.  The BLF did not turn on for a good reason.  If you want to go with the percentages, it breaks down like this:

Less than 1% light is out.  Check it anyway in diagnostics.  It does not take any time.

Less than 1%The phone is broken.  This is very unlikely for a BLF

85%Cabling – Loose connector – bad patch cord

13%QOS not set up or set up wrong

Moral of the story:

Carry a known good patch cable and some tools to re-connect the wall jacks.  Rule this out first and you will save a lot of time over the long run.

=== '''I am using a Plantronics headset and experience buzzing at my phone.''' ===

 Dealers have found that installing a PLANTRONICS 70584-01 RFI Filter resolved this issue.

=== 3 Way Conference alternate functionality ===

Sometimes an end user will want the ability to create a 3-way conference call at a phone, hang up, but have the other parties remain connected. This can be accomplished by modifying the Global Template for the phones, thereby modifying the behavior in each phone:

#Navigate in the PBX to PBX Setup=>Phone Global
#Click the button to Edit Current Template
#Find <code>ConfEndMethod="0"</code>
#Change the 0 to a 1
#Save the Template
#Apply Changes
#The next time the phone downloads it config (either on reboot or at 1am), the phone will have the new functionality.

Alternatively, you could log into a phone on a case by case basis, navigate to Advanced=>Phone Settings, and change the setting for Conference Exit Result: from Disconnect All to Others Remain Connected. You will need to disable AutoDL config if you make this change manually, otherwise at 1am it will get its config from the PBX and undo your manual changes.

=== Locked up Phone Recovery - Defaulting without a Network Connection ===

We've found that in some instances a phone may download an improperly formatted config file. This could result in sporadic symptoms of the phone becoming locked or unresponsive, being unable to press any keys, etc.  In this state the phone will often display the time but not respond to key presses. If this occurs, try the following to resolve:

#Disconnect the phone entirely from the network
#Power via brick
#Default the phone
##Press Menu
##Choose Settings
##Choose Advanced Settings
##Press OK (no password)
##Select Load Default Settings
##Choose OK
##Reboot phone
#If the phone was supposed to be upgrading to a newer firmware version when the phone entered the locked up state then perform the following operations on the newly defaulted phone.
##Reconnect Phone to Network
##Autodiscover + Configure and Restart Phone
##Wait 5 minutes after phone appears to be registered it is important that you do not make any calls during this time as the phone is repairing the corrupted files.  Note that phone may not display upgrading firmware for 1-2 minutes.
##After phone upgrades it's firmware it should reboot automatically and be usable immediately.

=== The extension name is displaying incorrectly on phones since the name of that extension was changed, the Enterprise Phonebook does not seem to be updating ===

#Navigate to PBX Setup=>Phone Global and enable Enterprise Phonebook (Download and Generate).
#Go to PBX Setup=>Services and click the blue link TFTP Files.
#Find and delete enterprise_phonebook.xml, then Apply Changes.
#When Apply Changes completes, reboot the phones that are displaying the wrong names. This should resolve your issue.

=== How do I make the second call ringing to my phone get my Busy VM greeting instead of my NoAnswer greeting? ===

*One method to do this would be to have the user press the Reject key on their phone when the second call comes in.
*Alternatively, you can log into the IP address of the phone, navigate to Phone Settings=>Basic and set Call Waiting to Off. In this configuration the phone will auto DND a second incoming call while you are on the phone. This causes the busy greeting to play any time a second call is sent to the phone. NOTE: you will need to turn off automatic configuration updates in any phones that you want to have this setting on or change it globally in the phone configuration template.

=== Side Tone ===

Side tone is an old TDM feature where you hear yourself back when you speak. We do not advise to enable this, as side tone can cause issues with SIP If your end user insists on having side tone, here is how you add it to the default template instead of having to manually enable it in each phone. Rev 1 phones will not support this at all, so they will need Rev 2 or later phones.

*Edit the Default Phone Template found under PBX Setup=>Phone Global
*Find the section pertaining to SIP DSPs (this should be near the top)
[[File:SideTone.jpg|File:SideTone.jpg]]
*Add in <code>SideTone="1" </code> to this section. A setting of 1 means enabled, a setting of 0 means disabled.
*Save, Apply Changes, and then reboot the phones so they can download their updated config.

=== I accidentally entered http:// in the SIP Server field and now my phone is locked up ===

If you are manually configuring a phone and have accidentally entered http:// in the SIP server field, followed by an IP address, the phone will lock up. The display will show Xs and you will be unable to access the phone's WebGUI. If this has happened, we have an easily method to recover the phone.

*Disconnect the LAN cable and power up with a power brick. If already on a brick, reboot after removing the cable.
*Once booted, default the phone via the phone menu.

=== How do I access the Multiple Key Pages on the new IP620 ===

(Does not work on IP620PE)

To program the extra pages, you need to add an Expansion Module through the PBX programming. 32 keys = 4 pages of 8. Once programmed, you will use the left and right directional key on the phone to navigate to the extra pages. An indicator at the bottom of the screen will show you what pages is displaying.

::NOTE:: There is currently a known issue where after the phone has downloaded its config it needs a reboot in order to display the new pages of keys.

 [[File:IP620 screen view.gif|File:IP620 screen view.gif]]

=== How do I use the EHS/Bluetooth Dongle function on the new IP620B and IP320 rev2.3.0 ===

EHS functionality will require an EHS module that plugs into the back of the phone. Once installed, you can use a standard wireless or wireless Bluetooth headset. The following guide will go over installation.

[[File:IPitomy-BWM36 EHS Configuration User Guide.pdf|File:IPitomy-BWM36 EHS Configuration User Guide.pdf]]

Once installed and paired, calls will ring to the headset.  To answer, press the button on the headset once.  To hang up, press the button on the headset once.

NOTE: The phones need to be on version 2.3.2.4 or greater for the EHS to work.

A note about Bluetooth headsets with a UC Base unit:

[[File:Bluetooth pairing UC.pdf|File:Bluetooth pairing UC.pdf]]

Here are some models of headset reported to work by dealers or used here in our office:

*Delton X10
*BackBeat 903
*Plantronics Voyager Pro 510
*Plantronics PLT- D655
*Motorola 537X
*Plantronics M100
*Plantronics Discovery 600
*Inland ProHT Bluetooth Headset
*Jabra 920 (be sure to disable Missed Call in the headset interface or it will play a tone that sounds like ringing when you miss a call)
*[http://www.amazon.com/Bluetooth-Microphone-Headphone-Rechargable-Cancelling/dp/B013QV6UZ8/ref=sr_1_64?s=wireless&ie=UTF8&qid=1446722675&sr=1-64&keywords=AGPtek http://www.amazon.com/Bluetooth-Microphone-Headphone-Rechargable-Cancelling/dp/B013QV6UZ8/ref=sr_1_64?s=wireless&ie=UTF8&qid=1446722675&sr=1-64&keywords=AGPtek]
*[http://www.amazon.com/gp/product/B008HLVWH4?psc=1&redirect=true&ref_=oh_aui_search_detailpage http://www.amazon.com/gp/product/B008HLVWH4?psc=1&redirect=true&ref_=oh_aui_search_detailpage]

 

=== How do I load a new background image on the IP620BG? ===

You will need to log into the phone via its IP address. Once there, navigate to Phone Settings=>Wallpaper. The standard idle background in the phone is the Standby image, and must be a file that is 272x384.

If you have loaded a bad image file and want to return to the background shipped with the phone, here is the file:

[[File:IPitomy standby.jpg|File:IPitomy standby.jpg]]

=== IP290 Wall Mount Solution ===

If you would like to wall mount your IP290 phone, the following guide will give a functional option. The switch plate needed is available at most big box stores.

[[File:290 wall mount.pdf|File:290 wall mount.pdf]]

=== General Troubleshooting ===

*The key to isolating new issues is to change one thing at a time and see what fixes the issue. The possible causes are 1) Hardware defect 2) Configuration defect 3) Network Issue 4) Software defect

*What is the background / history of the issue? For example: This phone is Bob the IT guy's home phone. It has worked for 6 months without an issue, and now we are seeing an issue. Or we have done an install and everything is working except one phone?

*Isolate the Issue. The easiest way to do this is to swap the phone with one that is working fine and has not demonstrated the issue. If the problem follows the phone, it is most likely a configuration or hardware issue. If the problem happens on the other phone in this same location, the problem is probably on the segment of the LAN. After this has been tested, try swapping the ports on the data switch. It could be a bad port on the switch. Another option is to swap cabling, or switch from POE to a power brick for testing.

*Ensure the phone is on current firmware. If you have two phones on the same firmware, and only one experiences the issue, we can fairly confidently rule out a software defect.

*Try defaulting and reprogramming the phone, this should eliminate any configuration issues.

*Once all this is done, we would have found the cause of the problem or eliminated the majority of possible issues. At this time we would investigate the network. [[Troubleshooting Network]]

HD Phone FAQ

2015-12-22T18:05:47Z

John Wolfe: /* How do I use the EHS/Bluetooth Dongle function on the new IP620B and IP320 rev2.3.0 */

__NOTITLE__

== HD Phone FAQ ==

=== I have an IP330 on a gigabit switch and I am periodically having issues with connectivity from a PC connected through the phone. Is there anything I can do to resolve the issue? ===

On some older Network cards we have had reports of trouble when they go to sleep. This is apparently a common problem with an older RealTek network cards.

Here are the things that we have seen work in the field:

*Replace the cable with a gigabit cable. There are greater requirements upon the quality of the cable when supporting gigabit.
*In the hardware properties of the network card in windows, disable sleep. (In Win7 this is under Power Management -> Allow computer to turn off this device to save power).
*Set the speed to a fixed value in the PC, try 1000 or 10/100

=== How can I download logs from the HD phone? ===

Sometimes IPitomy will require you to get the logs from your phone so we can investigate an issue more deeply. Steps to accomplish this are:

#Log into the IP address of the phone (username: root/password: root)
#Navigate to Phone Maintenance=>HTTP Upgrade
#Click the Download button next to All Config File:

NOTE:It may take a few minutes to download all of the files, be patient.

=== How do I configure my my HD Phones to use VLANs? ===

If you need to use VLAN to control what IP addresses the phones pull from DHCP you can modify the Phone Configuration Template in the PBX. VLANs logically isolate network traffic, they do not necessarily physically isolate it. It is typically going to be used because there are more devices on the network then there are IP Addresses in a single subnet. Or in order to implement security.

#Navigate to PBX Setup=>Phone Global
#Click Edit Current Template
#Find the following section in the template pertains to VLAN, this is how it appears by default

<code>
<VLANs>
<Vlan LocalEnableVlan="0" PCEnableVlan="0" LocalVID="0" PCVID="0" LocalPriority="0" PCPriority="0" EnableVlan="0"/>
</VLANs>
</code>

{| class="wikitable"
|-
! Variable
! Function
! Settings
|-
| LocalEnableVlan
| Controls if the LAN port is using vLAN tagging
| 0 = Disabled, 1 = Enabled
|-
| PCEnableVlan
| Controls if the PC port is using vLAN tagging
| 0 = Disabled, 1 = Enabled
|-
| LocalVID
| Defines what vLAN tag the LAN port is using
| Set to the vLAN ID
|-
| PCVID
| Defines what vLAN tag the PC port is using
| Set to the vLAN ID
|-
| LocalPriority
| Not Used
| Leave at 0
|-
| PC Priority
| Not Used
| Leave at 0
|-
| EnableVlan
| Global setting to toggle vLAN on PC and LAN port
| 0 = Disabled, 1 = Enabled
|}

*In the following example the LAN port will use VLAN 5 and the PC port will use VLAN 8, allowing the Phone and PC to get different IP addresses from the DHCP server:

<VLANs>
<Vlan LocalEnableVlan="1" PCEnableVlan="1" LocalVID="5" PCVID="8" LocalPriority="0" PCPriority="0" EnableVlan="1"/>
</VLANs>

*When done modifying the template Save, Apply Changes, and now any phones that are configured to the PBX will have their configuration file modified to match the template changes.

NOTE: If the phones are default, you may have to statically set them to use the correct VLAN so they can communicate with the PBX to get their configuration.

=== How do I Enabled LLDP Globally for the Phones ===

In the Global Template (PBX Setup=>Phone Global) you can search for LLDP and will see <code>LLDP="0"</code>, this designates that LLDP is disabled. In order to enable you will need to change the 0 to a 1. You will also need to add the following <code> LLDPInterval="60" </code>.

When all is set the entry in the Global Template will look like this:
<code>LLDP="1" LLDPInterval="60"</code>

Click this link for general instructions on setting up [[LLDP]] in the switch.

=== How can I set my Phone to use a static IP address? ===

IPitomy supports configuring phones using DHCP and this is the best option for almost every network. If you would like to statically set IP addresses, we recommend that you use DHCP reservations to do this. In the vast majority of routers and DHCP servers there is an ability to assign a static IP address to a MAC address. While this process of assigning may be tedious, it is no less so than going to each individual phone and manually configuring the IP address (Static IP). In the end it eliminates any difficult modifications that you will need to make to configuration files and is just a way easier method to use for accomplishing the same task.

If having this information, the phone still must be manually set to static IP addresses then you will need to understand the following information. You have several options for performing this configuration:

==== Configure then Disable Configuration Updates ====

#Automatically Configure the phones using DHCP
#Go to phone's web page (root/root to log in) Navigate to Phone Maintenance->Auto Provision and uncheck the box that says Auto Download Config File.
#set the IP address to the desired static address
#The phone should keep it's original configuration and any changes made to phone configuration will now have to be done through the phone web interface. The exception to this is that you can change the phone keys in this configuration, just do NOT press Save and Configure. Press Save then Apply changes. The phone will update the keys at night when it downloads it's key configuration.

==== Change Configuration File Template ====

'''Please read the section of the Documentation [[http://wiki.ipitomy.com/index.php/HD_Phones#A_Note_on_Changing_the_Global_Template Note on Changing the Global Template]] to understand the implications of using this method.'''

The second option is more difficult to initially configure but is easier because you can still change phone settings. You should only do this if you are familiar with XML as you can break the configuration files by making changes here that lead to invalid XML.

#Go to PBXSetup->Phone Global
#Under "HD Phone Configuration Template" click "Reset Template to Default"
#Now click Edit Template. Under Networks - delete the entire Node called "network"
#Now phones should keep whatever you manually set them to.
#Note with this method that when you upgrade the PBX software you should be aware that we may make changes to the default template and you may have to reset the template and remake the changes in order to support additional functionality that is added in future releases.

The final option here is to just manually configure the phone. This works but if you call tech support with an issue they will probably have you default and reconfigure the phone through the PBX as we change a number of options when we provision the phones to the PBX.

=== Can I use HD Phones with my IP1000? ===

''YES.  HD phones will function normally as most other SIP phones with your IP1000. ''

In order to configure the phone you must use [[HD Phones#Manual|Manual Phone Configuration]].

=== My BLF won’t light on my phone? ===

If you have a phone where a BLF will not come on when it is supposed to, there are several things you can do.  The most important thing is to understand is how a BLF is turned on or off.  When a phone goes off hook and connects with a call, it sends out a short status packet to the PBX.  The packet simply says that the status has changed from available to busy.  The PBX then updates all of the phones that have a subscription to the BLF with a status packet indicating that the status has changed to busy.

The packet ether got there or it didn’t.  If it didn’t, the status will not update.  The status should update automatically every 30 minutes or so.  If the packet was missed, everything should be back to normal the next time the phone re-subscribes (about every 30 minutes automatically).  

What can go wrong?

The packet might have been lost.  What do I do? Why would the packet get lost in the first place?  Is it magic?  Is it being controlled by “The Force”?  Probably not.  The packet may have not arrived due to many factors.  Most likely the flow of normal packets was disrupted.  In the world of SIP, signaling packets have a very short life.  They become stale if they are not delivered on time.  The SIP packets are delivered using UDP.  This protocol does not ask for a positive acknowledgement when a packet is received.  It is assumed that it has been received.  There are several things that can cause a packet to be delayed on a LAN.  All these things have the same symptom.  

Here is a short troubleshooting guide:

#Perhaps you have recently restarted all services or rebooted your PBX.  You phones lose their BLF subscriptions when this occurs.  The phones will resubscribe automatically in time.  If you wish to have the BLFs work immediately then you will need to reboot the effected phones.
#The packet might not be the culprit; maybe your lamp is broken.  Test this by going to the button on the phone and select diagnostics.  Select the option lights.  Did the light come on?  If it did, it’s not the light.
#To narrow this down a little further, it is best to remove all doubt.  Restore the phone to factory defaults and reprogram it.  Test it and see if the problem happens again.  If it does, go to the next step.
#Troubleshoot the network.  See this guide for assistance [[Troubleshooting: Network|Network Troubleshooting]] 

There is no such thing as black magic.  The BLF did not turn on for a good reason.  If you want to go with the percentages, it breaks down like this:

Less than 1% light is out.  Check it anyway in diagnostics.  It does not take any time.

Less than 1%The phone is broken.  This is very unlikely for a BLF

85%Cabling – Loose connector – bad patch cord

13%QOS not set up or set up wrong

Moral of the story:

Carry a known good patch cable and some tools to re-connect the wall jacks.  Rule this out first and you will save a lot of time over the long run.

=== '''I am using a Plantronics headset and experience buzzing at my phone.''' ===

 Dealers have found that installing a PLANTRONICS 70584-01 RFI Filter resolved this issue.

=== 3 Way Conference alternate functionality ===

Sometimes an end user will want the ability to create a 3-way conference call at a phone, hang up, but have the other parties remain connected. This can be accomplished by modifying the Global Template for the phones, thereby modifying the behavior in each phone:

#Navigate in the PBX to PBX Setup=>Phone Global
#Click the button to Edit Current Template
#Find <code>ConfEndMethod="0"</code>
#Change the 0 to a 1
#Save the Template
#Apply Changes
#The next time the phone downloads it config (either on reboot or at 1am), the phone will have the new functionality.

Alternatively, you could log into a phone on a case by case basis, navigate to Advanced=>Phone Settings, and change the setting for Conference Exit Result: from Disconnect All to Others Remain Connected. You will need to disable AutoDL config if you make this change manually, otherwise at 1am it will get its config from the PBX and undo your manual changes.

=== Locked up Phone Recovery - Defaulting without a Network Connection ===

We've found that in some instances a phone may download an improperly formatted config file. This could result in sporadic symptoms of the phone becoming locked or unresponsive, being unable to press any keys, etc.  In this state the phone will often display the time but not respond to key presses. If this occurs, try the following to resolve:

#Disconnect the phone entirely from the network
#Power via brick
#Default the phone
##Press Menu
##Choose Settings
##Choose Advanced Settings
##Press OK (no password)
##Select Load Default Settings
##Choose OK
##Reboot phone
#If the phone was supposed to be upgrading to a newer firmware version when the phone entered the locked up state then perform the following operations on the newly defaulted phone.
##Reconnect Phone to Network
##Autodiscover + Configure and Restart Phone
##Wait 5 minutes after phone appears to be registered it is important that you do not make any calls during this time as the phone is repairing the corrupted files.  Note that phone may not display upgrading firmware for 1-2 minutes.
##After phone upgrades it's firmware it should reboot automatically and be usable immediately.

=== The extension name is displaying incorrectly on phones since the name of that extension was changed, the Enterprise Phonebook does not seem to be updating ===

#Navigate to PBX Setup=>Phone Global and enable Enterprise Phonebook (Download and Generate).
#Go to PBX Setup=>Services and click the blue link TFTP Files.
#Find and delete enterprise_phonebook.xml, then Apply Changes.
#When Apply Changes completes, reboot the phones that are displaying the wrong names. This should resolve your issue.

=== How do I make the second call ringing to my phone get my Busy VM greeting instead of my NoAnswer greeting? ===

*One method to do this would be to have the user press the Reject key on their phone when the second call comes in.
*Alternatively, you can log into the IP address of the phone, navigate to Phone Settings=>Basic and set Call Waiting to Off. In this configuration the phone will auto DND a second incoming call while you are on the phone. This causes the busy greeting to play any time a second call is sent to the phone. NOTE: you will need to turn off automatic configuration updates in any phones that you want to have this setting on or change it globally in the phone configuration template.

=== Side Tone ===

Side tone is an old TDM feature where you hear yourself back when you speak. We do not advise to enable this, as side tone can cause issues with SIP If your end user insists on having side tone, here is how you add it to the default template instead of having to manually enable it in each phone. Rev 1 phones will not support this at all, so they will need Rev 2 or later phones.

*Edit the Default Phone Template found under PBX Setup=>Phone Global
*Find the section pertaining to SIP DSPs (this should be near the top)
[[File:SideTone.jpg|File:SideTone.jpg]]
*Add in <code>SideTone="1" </code> to this section. A setting of 1 means enabled, a setting of 0 means disabled.
*Save, Apply Changes, and then reboot the phones so they can download their updated config.

=== I accidentally entered http:// in the SIP Server field and now my phone is locked up ===

If you are manually configuring a phone and have accidentally entered http:// in the SIP server field, followed by an IP address, the phone will lock up. The display will show Xs and you will be unable to access the phone's WebGUI. If this has happened, we have an easily method to recover the phone.

*Disconnect the LAN cable and power up with a power brick. If already on a brick, reboot after removing the cable.
*Once booted, default the phone via the phone menu.

=== How do I access the Multiple Key Pages on the new IP620 ===

(Does not work on IP620PE)

To program the extra pages, you need to add an Expansion Module through the PBX programming. 32 keys = 4 pages of 8. Once programmed, you will use the left and right directional key on the phone to navigate to the extra pages. An indicator at the bottom of the screen will show you what pages is displaying.

::NOTE:: There is currently a known issue where after the phone has downloaded its config it needs a reboot in order to display the new pages of keys.

 [[File:IP620 screen view.gif|File:IP620 screen view.gif]]

=== How do I use the EHS/Bluetooth Dongle function on the new IP620B and IP320 rev2.3.0 ===

EHS functionality will require an EHS module that plugs into the back of the phone. Once installed, you can use a standard wireless or wireless Bluetooth headset. The following guide will go over installation.

[[File:IPitomy-BWM36 EHS Configuration User Guide.pdf|File:IPitomy-BWM36 EHS Configuration User Guide.pdf]]

Once installed and paired, calls will ring to the headset.  To answer, press the button on the headset once.  To hang up, press the button on the headset once.

NOTE: The phones need to be on version 2.3.2.4 or greater for the EHS to work.

A note about Bluetooth headsets with a UC Base unit:

[[File:Bluetooth pairing UC.pdf|File:Bluetooth pairing UC.pdf]]

Here are some models of headset reported to work by dealers or used here in our office:

*Delton X10
*BackBeat 903
*Plantronics Voyager Pro 510
*Plantronics PLT- D655
*Motorola 537X
*Plantronics M100
*Plantronics Discovery 600
*Inland ProHT Bluetooth Headset
*Jabra 920 (be sure to disable Missed Call in the headset interface or it will play a tone that sounds like ringing when you miss a call)
*[http://www.amazon.com/Bluetooth-Microphone-Headphone-Rechargable-Cancelling/dp/B013QV6UZ8/ref=sr_1_64?s=wireless&ie=UTF8&qid=1446722675&sr=1-64&keywords=AGPtek http://www.amazon.com/Bluetooth-Microphone-Headphone-Rechargable-Cancelling/dp/B013QV6UZ8/ref=sr_1_64?s=wireless&ie=UTF8&qid=1446722675&sr=1-64&keywords=AGPtek]
*[http://www.amazon.com/gp/product/B008HLVWH4?psc=1&redirect=true&ref_=oh_aui_search_detailpage http://www.amazon.com/gp/product/B008HLVWH4?psc=1&redirect=true&ref_=oh_aui_search_detailpage]

'''Configuration of Plantronics CS540'''

To use the Plantronics CS540 with the EHS device it is important to set it up according to the following diagrams.

[[File:1511.JPG|File:1511.JPG]]

The setting on the base unit should be dialed in as set in this photo.

[[File:1512.JPG|File:1512.JPG]]

The Cable shown above is the cable used to connect to the EHS

[[File:1513.JPG|File:1513.JPG]]

Connect as shown in the folloing two photos.

[[File:1514.JPG|File:1514.JPG]]

=== How do I load a new background image on the IP620BG? ===

You will need to log into the phone via its IP address. Once there, navigate to Phone Settings=>Wallpaper. The standard idle background in the phone is the Standby image, and must be a file that is 272x384.

If you have loaded a bad image file and want to return to the background shipped with the phone, here is the file:

[[File:IPitomy standby.jpg|File:IPitomy standby.jpg]]

=== IP290 Wall Mount Solution ===

If you would like to wall mount your IP290 phone, the following guide will give a functional option. The switch plate needed is available at most big box stores.

[[File:290 wall mount.pdf|File:290 wall mount.pdf]]

=== General Troubleshooting ===

*The key to isolating new issues is to change one thing at a time and see what fixes the issue. The possible causes are 1) Hardware defect 2) Configuration defect 3) Network Issue 4) Software defect

*What is the background / history of the issue? For example: This phone is Bob the IT guy's home phone. It has worked for 6 months without an issue, and now we are seeing an issue. Or we have done an install and everything is working except one phone?

*Isolate the Issue. The easiest way to do this is to swap the phone with one that is working fine and has not demonstrated the issue. If the problem follows the phone, it is most likely a configuration or hardware issue. If the problem happens on the other phone in this same location, the problem is probably on the segment of the LAN. After this has been tested, try swapping the ports on the data switch. It could be a bad port on the switch. Another option is to swap cabling, or switch from POE to a power brick for testing.

*Ensure the phone is on current firmware. If you have two phones on the same firmware, and only one experiences the issue, we can fairly confidently rule out a software defect.

*Try defaulting and reprogramming the phone, this should eliminate any configuration issues.

*Once all this is done, we would have found the cause of the problem or eliminated the majority of possible issues. At this time we would investigate the network. [[Troubleshooting Network]]

File:1514.JPG

2015-12-22T18:05:09Z

John Wolfe:

{{RMImage
|Filename=File:1514.JPG
|CopyrightBy=
|CreationDate=2015/12/22
|Uploader=John Wolfe
|UploadDate=2015/12/22
|Description=
|Keywords=
|Format=image/jpeg
|RelatedArticles=HD Phone FAQ
}}

File:1514.JPG

2015-12-22T18:05:09Z

John Wolfe:

File:1513.JPG

2015-12-22T18:04:19Z

John Wolfe:

{{RMImage
|Filename=File:1513.JPG
|CopyrightBy=
|CreationDate=2015/12/22
|Uploader=John Wolfe
|UploadDate=2015/12/22
|Description=
|Keywords=
|Format=image/jpeg
|RelatedArticles=HD Phone FAQ
}}

File:1513.JPG

2015-12-22T18:04:18Z

John Wolfe:

File:1512.JPG

2015-12-22T18:03:32Z

John Wolfe:

{{RMImage
|Filename=File:1512.JPG
|CreationDate=2015/12/22
|Uploader=John Wolfe
|UploadDate=2015/12/22
|Format=image/jpeg
|RelatedArticles=HD Phone FAQ
}}

File:1512.JPG

2015-12-22T18:03:32Z

John Wolfe: uploaded a new version of "File:1512.JPG"

{{RMImage
|Filename=File:1512.JPG
|CopyrightBy=
|CreationDate=2015/12/22
|Uploader=John Wolfe
|UploadDate=2015/12/22
|Description=
|Keywords=
|Format=image/jpeg
|RelatedArticles=HD Phone FAQ
}}

File:1512.JPG

2015-12-22T18:02:46Z

John Wolfe:

{{RMImage
|Filename=File:1512.JPG
|CopyrightBy=
|CreationDate=2015/12/22
|Uploader=John Wolfe
|UploadDate=2015/12/22
|Description=
|Keywords=
|Format=image/jpeg
|RelatedArticles=HD Phone FAQ
}}

File:1512.JPG

2015-12-22T18:02:46Z

John Wolfe:

File:1511.JPG

2015-12-22T18:01:37Z

John Wolfe:

{{RMImage
|Filename=File:1511.JPG
|CopyrightBy=
|CreationDate=2015/12/22
|Uploader=John Wolfe
|UploadDate=2015/12/22
|Description=
|Keywords=
|Format=image/jpeg
|RelatedArticles=HD Phone FAQ
}}

File:1511.JPG

2015-12-22T18:01:37Z

John Wolfe:

Mitel 5224 Setup

2015-10-30T14:44:29Z

John Wolfe: Created page with "# Confiture an extension in '''IPitomy | destinations| Extensions |''' with the extension '''number''' configured in '''general''' settings and, in '''advanced''' settings, t..."

#

Confiture an extension in '''IPitomy | destinations| Extensions |''' with the extension '''number''' configured in '''general''' settings and, in '''advanced''' settings, the '''password '''and'''phone type''' ('''Generic''') the rest should be default

#

On the Mitel 5224:

#

Factory default

a. Press '''both''' the '''up''' and '''down''' keys while applying power to the phone

b. At '''Network settings''' press '''No''' until '''Tools & Features''' appear

c. Press '''Yes'''

d. Press '''No''' until '''Restore Defaults''' appear

e. Press '''Yes'''

f. Press '''Confirm'''

#

SIP Mode

a. Press the '''Super key''' (blue key) while applying power to the phone

b. At '''Configure Phone''' press '''Ye'''s

c. Press '''No''' until '''Phone Mode''' appears, Press '''Yes'''

d. At '''Protocol''' press '''Yes'''

e. At '''Phone Mode Minet''' press '''change''' then press '''SIP'''

f. At '''Store Changes''' press '''Yes'''

g. At '''reboot now''' press '''yes'''

#

Access Web GUI for 5224

a. When phone is booted up press the S'''uper key'''

b. Press the bottom right feature key on phone set, IP address and MAC address will appear

c. Open a web browser and go to the IP address from b. above

d. A Logon Window will appear, the '''User Nam'''e is '''admin''' the '''pasword''' is '''5224'''

e. The following screen appears on login

f. The left hand column has links to specific programming pages as well as the firmware load at the top of the page. The firmware on this phone was 04.00.00.07 features and methods may vary with different firmware loads.

#

Network Config

a. Click the '''Network Config''' Link

b. Set the '''Basic''' | '''SIP Phone Host Name'''

c. Set the '''Domain Name''' to the IP of the ipitomy IPPBX

d. Set '''DHCP''' as desired (dhcp on/off)

e. Set the '''SIP Phone IP Address , Subnet Mask, Default Gateway''' and '''DNS''' as needed (with DHCP '''On,''' this will display the DHCP addresses.

f. Set the '''Additional Servers | SNTP Server''' to the ip of the Ipitomy IPPBX

g. Set the '''Time Zone , Daylight Savings Mode and Start/End Settings''' as desired

h. Click '''Save and Reboot''' or '''apply''' if more programming is to be done

#

User config

a. Click the '''User Config''' link

b. Insert Extension number in '''User ID and SIP Authentication User Name'''fields

c. Insert the name to be displayed in the 5224 display in the U'''ser Display Name''' field

d. Insert the SIP authentication password from the ipitomy D'''estination|Extensions|Advanced p'''age in the'''SIP Authentication Password '''field

e. Click S'''ave and Reboot '''or'''apply''' if more programming is to be done

#

SIP config

a. Click the '''SIP Config''' link on left side

b. Insert the IP address of the Ipitomy IPPBX in the: '''SIP Proxy server, Sip Registry Server, Outbound Server URL, '''and'''Voicemail Server''' fields

c. Change '''Authenticate Mode''' to '''Basic'''

d. Click '''Save and Reboot '''or'''Apply''' if more programming is to be done

#

Media Config

a. click the '''Media Config''' link

b. set '''DTMF type''' to '''automatic'''

c. set '''Media Start Port''' to '''10000'''

d. set '''Media End Port''' to '''20000'''

e. click S'''ave and Reboot '''or'''Apply''' if more programming is to be done

Note: the mitel should link to the IPitomy at this time with a S'''ave and Reboot'''. On the mitel

<ol start="10">
<li>Key Programming the first 4 keys on bottom right of the phone are pre-assigned and cannot be changed. The top 3 keys on top right are pre-assigned and may be changed. Active call features cannot be programmed onto feature keys due to the key will select a new line key to make the new speed dial call and not affect the in progress call. a Click the '''Key Programming''' Link</li>
</ol>

b. click on the P'''rogrammable Key:''' field and select a button

c. click on the '''Feature''' and select from the drop down list of features, the only features recommended to be used are '''Speed dial key, Call log, Advisory message on/off, and Headset on/off.'''

d. If '''Speed Dial Key''' was selected click on the '''Speed Dial Address''' and enter the number desired to be dialed. Leave the '''Address Name/Number''' field at default.

e. Click '''Apply''' (not applying will result in programming not being saved) repeat above steps until all keys are programmed and then click '''Save and Reboot'''.

f. after reboot go back and verify the changes remained programmed in the phone.
<ol start="10">
<li></li>
<li>Date and Time a. Click the '''Date/Time''' field</li>
</ol>

b. set the correct date and time

c. Click '''Apply'''

File:Mitel 5224 setup for ipitomy.odt

2015-10-30T14:40:36Z

John Wolfe:

{{RMImage
|Filename=File:Mitel 5224 setup for ipitomy.odt
|CreationDate=2015/10/30
|Uploader=John Wolfe
|UploadDate=2015/10/30
|Format=application/vnd.oasis.opendocument.text
|RelatedArticles=Mitel 5224 Setup
}}

File:Mitel 5224 setup for ipitomy.odt

2015-10-30T14:40:35Z

John Wolfe: uploaded a new version of "File:Mitel 5224 setup for ipitomy.odt"

{{RMImage
|Filename=File:Mitel 5224 setup for ipitomy.odt
|CopyrightBy=
|CreationDate=2015/10/30
|Uploader=John Wolfe
|UploadDate=2015/10/30
|Description=
|Keywords=
|Format=application/vnd.oasis.opendocument.text
|RelatedArticles=Mitel 5224 Setup
}}

File:Mitel 5224 setup for ipitomy.odt

2015-10-30T14:39:32Z

John Wolfe:

File:Mitel 5224 setup for ipitomy.odt

2015-10-30T14:39:31Z

John Wolfe:

PRI

2015-10-26T15:35:51Z

John Wolfe:

== Headline text ==

== Multi Port/Card PRI Configuration ==

{{:Multi Port PRI Configuration}}

== Troubleshooting a PRI ==

#Always be sure to check that the settings for the card match what the provider wants. (Switch type is a big one).
#If you see HDLC Abort of Primary D-Chan, you most likely have a timing slip on the PRI. Replacing the cat-5 the provider usually uses with T1 individually shielded pair wiring often times helps with this, as well as potential echo on the lines.
#Ipitomy PRI/T1 cards always expect a crossover shielded T1 cable. The below diagram shows wiring pin outs to use.

#[[File:Image.png.png|File:Image.png.png]]

Note: It is often the case that the use of Standard Cat 5 Cabling becomes an issue. Standard Cat5 is sufficient for TCPIP networking. But the Specification for T1 Data networks is far different from TCPIP Data networks. The following Link is very descriptive and shows the original ATT specification. ALWAYS use a Individually shielded T1 Crossover Cable to connect from the demarq to the Ipitomy Card. [http://www.quabbin.com/page2027.html http://www.quabbin.com/page2027.html]

=== Echo ===

If you are experiencing echo on your PRI lines, here are a few things you can try modifying:

*LBO - By default LBO is set to 0. We have found its a good idea to try -7.5. If this does not work, contact the provider and see if they can tell you how far the dmark is and what to set the LBO to. If none of the settings work, set it back to 0.

*Cabling - Most of the time the provider will install the PRI with cat5 cabling. While this can work, we have found that installing T-1 Cabling (individually shielded pairs) has in the past alleviated echo issues. The reason why is that cat-5 is not really rated to push the voltage needed for a PRI, and without the pairs being shielded themselves, bleed over is possible.  You can overcome the shielding issues by running it as two CAT-5 cables.

*Echo Cancel Algorithm - Changing this from MG2 to OSLEC has helped eliminate echo in certain installs.

*Echo Training - Increasing this from the default 800 to 1200 has eliminated echo on installs in the past.

=== Caller ID ===

If you have no or partial caller id - change answer after to two seconds. To do this navigate to the Channel Group settings and in the field titled "Answer After" set this to two seconds (or more, but 2 is a good start).

If you have no Caller ID, make sure in the Channel Group parameters, you have Use CID set to yes, and also in the Inbound Caller ID field set this value to asreceived .

=== '''NOTE: On reconfiguring a PRI''' ===
Should you find it neccesary to reconfigure a PRI that has been in use, it is required before you reconfigure and remove the existing channel groups, that you remove these groups from Outbound Routing. Making sure to save and apply. At that point you should be able to remove the existing channel groups and reconfigure the new ones. The system will not allow you to remove channel groups which are listed as outbound routes.

Square Key Ipitomy

2015-09-22T21:13:33Z

John Wolfe:

= '''Square Key on an Ipitomy System''' =

Infrequently Ipitomy is asked is there a way to Square Key an Ipitomy or to basically allow there to be Line Appearances on a phone. While there is not a way to actually map a line to a key on a phone this solution does give the required result.

In this example you would be using 4 Extensions per Line Appearance and 4 Groups also. You would need to make an Extension for each Line Key on a phone. Additionally you would need to use the manual method for programming the phones in question. As you would need to turn off Auto Download Config in the phone and then manually enter the extension programming for each Line Key. So if you had for instance 4 lines, you would need to program these four extension under each of those line Keys. Then you would create four correspnding groups. You would direct ringing on these groups from the Line or DID in question. In each group there would be one of the four extensions. The call would progress as follows.

== '''Ringing on Line 1 => sent to Ring group for line one => sent to extension for Line one''' ==

This achieves the result of having a Line appearance which could be programmed on as many phones as you like. The only downside is the manual programming of the phones, and the additional Extension licenses required.

Square Key Ipitomy

2015-09-22T21:12:17Z

John Wolfe: Created page with "= '''Square Key on an Ipitomy System''' = Infrequently Ipitomy is asked is there a way to Square Key an Ipitomy or to basically allow there to be Line Appearances on a p..."

= '''Square Key on an Ipitomy System''' =

Infrequently Ipitomy is asked is there a way to Square Key an Ipitomy or to basically allow there to be Line Appearances on a phone. While there is not a way to actually map a line to a key on a phone this solution does give the required result.

In this example you would be using 4 Extensions per Line Appearance and 4 Groups also. You would need to make an Extension for each Line Key on a phone. Additionally you would need to use the manual method for programming the phones in question. As you would need to turn off Auto Download Config in the phone and then manually enter the extension programming for each Line Key. So if you had for instance 4 lines, you would need to program these four extension under each of those line Keys. Then you would create four correspnding groups. YOu would direct ringing on these groups from the Line or DID in question. IN each group there would be one of the four extensions. The call would progress as follow.

== '''Ringing on Line 1 => sent to Ring group for line one => sent to extension for Line one''' ==

This achieves the result of having a Line appearance which could be programmed on as many phones as you like. The only downside is the manual programming of the phones, and the additional Extension licenses required.

Q Manager

2015-07-29T14:05:06Z

John Wolfe: /* MAC Users */

{{IP_PBX_Manual|sortkey=QManager}}

Q Manager is a Windows application that allows users to monitor and interact with calls. Your PBX must be on version 4.6.1 in order to use Q Manager, and 4.8.0+ to use the integrated chat feature. Licenses are sold to allow extensions access to the Q Manager feature.

Click the link for Q Manager [[http://wiki.ipitomy.com/wiki/Q_Manager_Release_Notes| Release Notes]]

=== MAC Users ===

If you have a user that has MAC and not PC, Q Manager software will not run. To accommodate this, we have created something we call Web manager. To access, the user will still need to have a license assigned to their extension for Q Manager, but they will access a webpage instead of running a piece of software. The address for Web Manager would be <PBXipAddress>/ippbx/WebManager. The user will be prompted to login, the username will be their extension number, and the password will be their voicemail PIN. Web Manager does not have all of the advanced functionality of Q Manager, but you are able to left click on a call and do some actions.

NOTE: The URL is case sensitive.

*Pickup: Use this to grab a call that is ringing on an extension other than yours.
*Page: Use this to page another extension.
*Call: Use this to call another extension.

Idle extensions will show as Green, Ringing extensions will show as Light Green, and extensions on a call will show as Red.

Parked Calls will show on the right side of the screen, clicking a parked call will bring up an option to Pickup, which retrieves that call from park.

[[File:WebManager.jpg|File:WebManager.jpg]]

== Installation ==

*Log into the PBX and assign Q Manager licenses to extensions. This is found under PBX Setup=>Services.
**User licenses have basic functionality, QManager licenses gain the ability to Barge and to Drag/Drop calls from Queue onto particular extensions.
*Save and Apply Changes.
*Click the Restart Call Manager button.
*Q Manager requires that your PC have .NET Framework 4.0 installed.
*Once .NET Framework 4.0 is installed, click the following link for the [[http://www.ipitomy.com/pbx_files/cm3publish/index.html Q Manager File]]
**NOTE: You may need to be using Internet Explorer to one click install.
*Click Settings in Q Manager if the settings are not displaying on launch
*Set Username to an Extension Number that is assigned a license
*Set Password to the Voicemail PIN of the Extension
*Set Server Address to the IP of the PBX
*Leave Port at the default of 5048
*Save and so long as the PC is in an IP range included in the ACL for Call Manager, you will have a working copy of Q Manager.

[[File:QManager-GeneralSettings.jpg|File:QManager-GeneralSettings.jpg]]

*NOTE: If you would like to use Q Manager as a remote user, we advise that the remote PC come in over a VPN.

==Training Video==
The following [[http://www.youtube.com/watch?v=ArDyehd6dXI| Video]] goes over how to use Q Manager, with some basic setup instructions near the end.

[[http://wiki.ipitomy.com/wiki/Q_Manager_Icons| Q Manager Icons Explained]]

== Chat ==

The PBX has a built in XMPP Server that allows Q Manager to use Chat.

*Under PBX Setup=>Chat, make sure Enable Chat Server is set to Yes
*Ensure that Chat Server Status says Running
*Apply Changes

[[File:QManager-ChatConfig.jpg|File:QManager-ChatConfig.jpg]]

*Click Settings in Q Manager
*Click Chat and ensure its enabled
**Q Manager automatically uses the same settings for the extension

NOTE: Chat must be enabled for a user to see an Abandoned Call Notification for a ring group.

[[File:QManager-ChatConfig2.jpg|File:QManager-ChatConfig2.jpg]]

IPitomy Communicator

2015-07-21T14:26:53Z

John Wolfe: /* Ipitomy Softphone */

Note: This Guide will assist the user in Installing and Using the IPitomy Communicator Softphone.

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!

The full commercial version is called IPitomy Communicator.

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 so it will use an IPitomy license instead of an open license.

Links for SoftPhone:

Demo version:
http://relay.ipitomy.com/softcall_demo/publish.htm

Full version – IPitomy Communicator:
http://relay.ipitomy.com/cloud_communicator/publish.htm

== '''Ipitomy Softphone''' ==

'''Installation'''

'''Installation Video'''

[https://www.youtube.com/watch?v=p5gwURMp4eo '''https://www.youtube.com/watch?v=p5gwURMp4eo''']

'''Park Button Video'''

[https://www.youtube.com/watch?v=dPCVXiR6kf8 '''https://www.youtube.com/watch?v=dPCVXiR6kf8''']

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.

'''Setup'''

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.

[[File:SoftPhone3.jpg|File:SoftPhone3.jpg]][[File:SoftPhone4.jpg|File:SoftPhone4.jpg]]

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

[[File:SoftPhone5.jpg|File:SoftPhone5.jpg]]

Select the Add Button.

[[File:SoftPhone6.jpg|File:SoftPhone6.jpg]]

Enter the Following

Display Name = Name of user

User Name = Extension Number

User Password = Voicemail Password

SIP Password = SIP Password for Extension

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)

All other information can be left at Default.

[[File:SoftPhone7.jpg|File:SoftPhone7.jpg]]

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

[[File:SoftPhone8.jpg|File:SoftPhone8.jpg]]

Click close and you should be returned to the User interface and your account should register to the PBX.

[[File:SoftPhone9.jpg|File:SoftPhone9.jpg]][[File:SoftPhone10.jpg|File:SoftPhone10.jpg]]

Your Softphone is now ready to be used.

'''Import Extensions from the PBX to Contacts.'''

To Import all Extension in the PBX into the Contact list select User List - Manage Users - Ipitomy and then Refresh

[[File:SoftPhone11.jpg|File:SoftPhone11.jpg]]

Select the PBX from the Host list

[[File:SoftPhone14.jpg|File:SoftPhone14.jpg]]

Click OK. The Softphone will now pull down an Extension list from the PBX which you configured.

[[File:SoftPhone13.jpg|File:SoftPhone13.jpg]]

Grandstream 410X Setup

2015-07-07T16:31:26Z

John Wolfe: /* Procedure—Configuration */

== Grandstream GXW 410X Gateway ==

 

Description

When application of the Grandstream GXW 410X is desired for connection of FXO circuits onto the IPitomy IPBX the following configuration information can be used as a guide to install the adjunct properly.

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.

[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]

[[File:Gstream1.JPG|File:Gstream1.JPG]]

=== Procedure—Connections ===

1. Make wiring connections as necessary to the FXO circuits required.

2. Make wiring connections to the LAN per the photo above – Connect the Grandstream

WAN port to an available LAN port of the network switch/router being used on site.

[[File:Gstream2.JPG|File:Gstream2.JPG]]

=== Procedure—Power Up and Login ===

1. Power up the unit and identify its assigned IP address. (Typically assigned from the DCHP server of the host router.)

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.

3. When the Grandstream page is accessed, input the password (admin at default) and navigate to the pages below making the changes as defined.

=== IPORTANT: Procedure Line Setup ===

On first login to the Grandstream, you will be brought to the following page.

[[File:GrandstreamCPT.jpg|File:GrandstreamCPT.jpg]]

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 caue 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.

NOTE: These tests should be performed at the site and with the lines that the unit is intended to be in service upon.

=== Procedure—Configuration ===

1. Navigate to the Networks Basic Setting page and input the data as in the example.

[[File:SIP3.jpg|File:SIP3.jpg]] 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.) 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. 4. Click the button. This saves information on this page before moving on. 5. Navigate to the FXO Lines page. Then Select Dialing on the left. 6. Change the Stage Method(1/2) Ch1-4:1; 7. Navigate to Settings and then select Channels Settings on the left. 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 & 3. Usually only one is necessary.This data field is the routing of the calls that are received on this FXO circuit (It is reccomended to use our example on this line which is ch1:1111;ch2:2222;ch3:3333;ch4:4444; This will break out the channels individually and allow the most flexibillity in use.  We have programmed the Grandstream to route calls that have been received to the SIP Server using Profile 1.

[[File:Gstream3.JPG|File:Gstream3.JPG]]

8.Click the button. This saves information on this page before moving on.

=== Procedure—Configuration Profile ===

[[File:SIP.jpg|File:SIP.jpg]]

1. Navigate to the Profile 1 page. 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. 3. Also be sure that the SIP Registration field is set to No. 4.Click the button. This saves information on this page before moving on. 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. 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; {x+ | [x*]+}

[[File:Sip2.jpg|File:Sip2.jpg]] to be able to send a * to outside IVR's 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

[[File:Gstream5.JPG|File:Gstream5.JPG]]

=== Procedure—Configuring the IPitomy IP PBX for Grandstream GXW 410x ===

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)

2. Click “Save Changes”

3. Then click on theApply Changes (upper right) to make these settings operational in the Ipitomy IPBX.

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.

 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.

=== Procedure—Configuring Optional Outbound Routing Methods ===

The Grandstream allows for two methods of outbound dialing at the same time.

Round-Robin (Linear Hunt)

 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

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.

1. Navigate to the Channels page

2. Go to the Port Scheduling Schema (Voip->PSTN) section and input the code or codes that you wish (99x)

== Remote Grandstream Configuration ==

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.

=== Grandstream Changes ===

#On the Profile page (profile 1 if that is the only one being used) set SIP Registration to YES
#Under the Channels page, configure one channel with login credentials:
#Set the SIP server in the Profile to the PBX Public IP.

[[File:Remote Device - Channels.jpg|File:Remote Device - Channels.jpg]]

=== PBX SIP Provider Changes ===

#Instead of entering an IP address for Host, set to “dynamic”
#Set Register and Authentication to YES
#Set Username and Name to the value entered as SIP User ID and Authenticate ID in the grandstream
#Set Secret to the value entered for Authen Password in the grandstream

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.

== Troubleshooting ==

=== Procedure—Troubleshooting—Outbound Dialing ===

If you having trouble dialing outbound make the following changes on the FXO Lines page:

 1. Tweak the Disconnect Threshold from 100 to 300ms.

 2. Tweak the Minimum Delay Before Dialing Out from 500 to 750ms.

=== Procedure—Troubleshooting—Call Quality ===

 If you are having call quality issues try the following changes on the Channels page:

 1.Set Silence Suppression from YES to NO.

 2. Set Echo Cancellation from YES to NO

=== Procedure—Troubleshooting—Call Volume ===

 If you are having issue with call volume try the following changes on the Channels page:

 1.Increase/Decrease Tx to PSTN Audio Gain by increments of 3 for issues with external party volume.

 2. Increase/Decrease Rx from PSTN Audio Gain by increments of 3 for issues with internal party volume

''' Procedure—Troubleshooting—Call Buzzing Noise'''

If you are having issues with a buzz heard prior to a Menu prompt; try upgrading the Grandstream firmware:

1. Navigate to Advanced Settings page

2. Ensure HTTP is selected for the method to upgrade

3. Set Firmware Server Path: to firmware.grandstream.com

4. Set Automatic Upgrade to YES

5. Set Allow DHCP Option 66 to override server to No Click Update at the bottom of the page

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

=== Procedure—Troubleshooting—Restore Factory Default ===

To Restore Factory Defaults:

1. While powered up, hold the recessed Reset button in for 7+ seconds

'''[[File:Gstream6.JPG|File:Gstream6.JPG]] '''

'''Procedure Setup of QOS'''

1. Access the IP of the Grandstream.

2. Navigate in the Grandstream Web Admin to Networks – Advanced Settings

3. Set the Layer 3 Diff Serv Value to 24 Which is CS3 or in Binary 011000

4. Submit your changes and reboot

 {{:Grandstream_FXO_FAQ}}

Grandstream 410X Setup

2015-06-09T17:55:19Z

John Wolfe:

== Grandstream GXW 410X Gateway ==

 

Description

When application of the Grandstream GXW 410X is desired for connection of FXO circuits onto the IPitomy IPBX the following configuration information can be used as a guide to install the adjunct properly.

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.

[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]

[[File:Gstream1.JPG|File:Gstream1.JPG]]

=== Procedure—Connections ===

1. Make wiring connections as necessary to the FXO circuits required.

2. Make wiring connections to the LAN per the photo above – Connect the Grandstream

WAN port to an available LAN port of the network switch/router being used on site.

[[File:Gstream2.JPG|File:Gstream2.JPG]]

=== Procedure—Power Up and Login ===

1. Power up the unit and identify its assigned IP address. (Typically assigned from the DCHP server of the host router.)

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.

3. When the Grandstream page is accessed, input the password (admin at default) and navigate to the pages below making the changes as defined.

=== IPORTANT: Procedure Line Setup ===

On first login to the Grandstream, you will be brought to the following page.

[[File:GrandstreamCPT.jpg|File:GrandstreamCPT.jpg]]

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 caue 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.

NOTE: These tests should be performed at the site and with the lines that the unit is intended to be in service upon.

=== Procedure—Configuration ===

1. Navigate to the Networks Basic Setting page and input the data as in the example.

[[File:SIP3.jpg|File:SIP3.jpg]] 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.) 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. 4. Click the button. This saves information on this page before moving on. 5. Navigate to the FXO Lines page. 6. Change the Stage Method(1/2) Ch1-4:1; 7. 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 & 3. Usually only one is necessary.This data field is the routing of the calls that are received on this FXO circuit (It is reccomended to use our example on this line which is ch1:1111;ch2:2222;ch3:3333;ch4:4444; This will break out the channels individually and allow the most flexibillity in use.  We have programmed the Grandstream to route calls that have been received to the SIP Server using Profile 1.

[[File:Gstream3.JPG|File:Gstream3.JPG]]

8.Click the button. This saves information on this page before moving on.

=== Procedure—Configuration Profile ===

[[File:SIP.jpg|File:SIP.jpg]]

1. Navigate to the Profile 1 page. 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. 3. Also be sure that the SIP Registration field is set to No. 4.Click the button. This saves information on this page before moving on. 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. 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; {x+ | [x*]+}

[[File:Sip2.jpg|File:Sip2.jpg]] to be able to send a * to outside IVR's 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

[[File:Gstream5.JPG|File:Gstream5.JPG]]

=== Procedure—Configuring the IPitomy IP PBX for Grandstream GXW 410x ===

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)

2. Click “Save Changes”

3. Then click on theApply Changes (upper right) to make these settings operational in the Ipitomy IPBX.

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.

 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.

=== Procedure—Configuring Optional Outbound Routing Methods ===

The Grandstream allows for two methods of outbound dialing at the same time.

Round-Robin (Linear Hunt)

 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

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.

1. Navigate to the Channels page

2. Go to the Port Scheduling Schema (Voip->PSTN) section and input the code or codes that you wish (99x)

== Remote Grandstream Configuration ==

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.

=== Grandstream Changes ===

#On the Profile page (profile 1 if that is the only one being used) set SIP Registration to YES
#Under the Channels page, configure one channel with login credentials:
#Set the SIP server in the Profile to the PBX Public IP.

[[File:Remote Device - Channels.jpg|File:Remote Device - Channels.jpg]]

=== PBX SIP Provider Changes ===

#Instead of entering an IP address for Host, set to “dynamic”
#Set Register and Authentication to YES
#Set Username and Name to the value entered as SIP User ID and Authenticate ID in the grandstream
#Set Secret to the value entered for Authen Password in the grandstream

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.

== Troubleshooting ==

=== Procedure—Troubleshooting—Outbound Dialing ===

If you having trouble dialing outbound make the following changes on the FXO Lines page:

 1. Tweak the Disconnect Threshold from 100 to 300ms.

 2. Tweak the Minimum Delay Before Dialing Out from 500 to 750ms.

=== Procedure—Troubleshooting—Call Quality ===

 If you are having call quality issues try the following changes on the Channels page:

 1.Set Silence Suppression from YES to NO.

 2. Set Echo Cancellation from YES to NO

=== Procedure—Troubleshooting—Call Volume ===

 If you are having issue with call volume try the following changes on the Channels page:

 1.Increase/Decrease Tx to PSTN Audio Gain by increments of 3 for issues with external party volume.

 2. Increase/Decrease Rx from PSTN Audio Gain by increments of 3 for issues with internal party volume

''' Procedure—Troubleshooting—Call Buzzing Noise'''

If you are having issues with a buzz heard prior to a Menu prompt; try upgrading the Grandstream firmware:

1. Navigate to Advanced Settings page

2. Ensure HTTP is selected for the method to upgrade

3. Set Firmware Server Path: to firmware.grandstream.com

4. Set Automatic Upgrade to YES

5. Set Allow DHCP Option 66 to override server to No Click Update at the bottom of the page

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

=== Procedure—Troubleshooting—Restore Factory Default ===

To Restore Factory Defaults:

1. While powered up, hold the recessed Reset button in for 7+ seconds

'''[[File:Gstream6.JPG|File:Gstream6.JPG]] '''

'''Procedure Setup of QOS'''

1. Access the IP of the Grandstream.

2. Navigate in the Grandstream Web Admin to Networks – Advanced Settings

3. Set the Layer 3 Diff Serv Value to 24 Which is CS3 or in Binary 011000

4. Submit your changes and reboot

 {{:Grandstream_FXO_FAQ}}

Grandstream 410X Setup

2015-06-09T17:53:38Z

John Wolfe:

== Grandstream GXW 410X Gateway ==

 

Description

When application of the Grandstream GXW 410X is desired for connection of FXO circuits onto the IPitomy IPBX the following configuration information can be used as a guide to install the adjunct properly.

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.

[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]

[[File:Gstream1.JPG|File:Gstream1.JPG]]

=== Procedure—Connections ===

1. Make wiring connections as necessary to the FXO circuits required.

2. Make wiring connections to the LAN per the photo above – Connect the Grandstream

WAN port to an available LAN port of the network switch/router being used on site.

[[File:Gstream2.JPG|File:Gstream2.JPG]]

=== Procedure—Power Up and Login ===

1. Power up the unit and identify its assigned IP address. (Typically assigned from the DCHP server of the host router.)

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.

3. When the Grandstream page is accessed, input the password (admin at default) and navigate to the pages below making the changes as defined.

=== IPORTANT: Procedure Line Setup ===

On first login to the Grandstream, you will be brought to the following page.

[[File:GrandstreamCPT.jpg|File:GrandstreamCPT.jpg]]

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 caue 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.

NOTE: These tests should be performed at the site and with the lines that the unit is intended to be in service upon.

=== Procedure—Configuration ===

1. Navigate to the Networks Basic Setting page and input the data as in the example.

[[File:SIP3.jpg|File:SIP3.jpg]] 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.) 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. 4. Click the button. This saves information on this page before moving on. 5. Navigate to the FXO Lines page. 6. Change the Stage Method(1/2) Ch1-4:1; 7. 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 & 3. Usually only one is necessary.This data field is the routing of the calls that are received on this FXO circuit (It is reccomended to use our example on this line which is ch1:1111;ch2:2222;ch3:3333;ch4:4444; This will break out the channels individually and allow the most flexibillity in use.  We have programmed the Grandstream to route calls that have been received to the SIP Server using Profile 1.

[[File:Gstream3.JPG|File:Gstream3.JPG]]

8.Click the button. This saves information on this page before moving on.

=== Procedure—Configuration Profile ===

[[File:SIP.jpg|File:SIP.jpg]]

1. Navigate to the Profile 1 page. 2. Here the SIP Server must be programmed. Set this to be the IP Address of the IPBX. In our example, the address is 192.168.2.18. 3. Also be sure that the SIP Registration field is set to No. 4.Click the button. This saves information on this page before moving on. 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. 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; {x+ | [x*]+}

[[File:Sip2.jpg|File:Sip2.jpg]] to be able to send a * to outside IVR's 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

[[File:Gstream5.JPG|File:Gstream5.JPG]]

=== Procedure—Configuring the IPitomy IP PBX for Grandstream GXW 410x ===

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)

2. Click “Save Changes”

3. Then click on theApply Changes (upper right) to make these settings operational in the Ipitomy IPBX.

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.

 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.

=== Procedure—Configuring Optional Outbound Routing Methods ===

The Grandstream allows for two methods of outbound dialing at the same time.

Round-Robin (Linear Hunt)

 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

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.

1. Navigate to the Channels page

2. Go to the Port Scheduling Schema (Voip->PSTN) section and input the code or codes that you wish (99x)

== Remote Grandstream Configuration ==

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.

=== Grandstream Changes ===

#On the Profile page (profile 1 if that is the only one being used) set SIP Registration to YES
#Under the Channels page, configure one channel with login credentials:
#Set the SIP server in the Profile to the PBX Public IP.

[[File:Remote Device - Channels.jpg|File:Remote Device - Channels.jpg]]

=== PBX SIP Provider Changes ===

#Instead of entering an IP address for Host, set to “dynamic”
#Set Register and Authentication to YES
#Set Username and Name to the value entered as SIP User ID and Authenticate ID in the grandstream
#Set Secret to the value entered for Authen Password in the grandstream

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.

== Troubleshooting ==

=== Procedure—Troubleshooting—Outbound Dialing ===

If you having trouble dialing outbound make the following changes on the FXO Lines page:

 1. Tweak the Disconnect Threshold from 100 to 300ms.

 2. Tweak the Minimum Delay Before Dialing Out from 500 to 750ms.

=== Procedure—Troubleshooting—Call Quality ===

 If you are having call quality issues try the following changes on the Channels page:

 1.Set Silence Suppression from YES to NO.

 2. Set Echo Cancellation from YES to NO

=== Procedure—Troubleshooting—Call Volume ===

 If you are having issue with call volume try the following changes on the Channels page:

 1.Increase/Decrease Tx to PSTN Audio Gain by increments of 3 for issues with external party volume.

 2. Increase/Decrease Rx from PSTN Audio Gain by increments of 3 for issues with internal party volume

''' Procedure—Troubleshooting—Call Buzzing Noise'''

If you are having issues with a buzz heard prior to a Menu prompt; try upgrading the Grandstream firmware:

1. Navigate to Advanced Settings page

2. Ensure HTTP is selected for the method to upgrade

3. Set Firmware Server Path: to firmware.grandstream.com

4. Set Automatic Upgrade to YES

5. Set Allow DHCP Option 66 to override server to No Click Update at the bottom of the page

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

=== Procedure—Troubleshooting—Restore Factory Default ===

To Restore Factory Defaults:

1. While powered up, hold the recessed Reset button in for 7+ seconds

'''[[File:Gstream6.JPG|File:Gstream6.JPG]] '''

'''Procedure Setup of QOS'''

1. Access the IP of the Grandstream.

2. Navigate in the Grandstream Web Admin to Networks – Advanced Settings

3. Set the Layer 3 Diff Serv Value to 24 Which is CS3 or in Binary 011000

4. Submit your changes and reboot

 {{:Grandstream_FXO_FAQ}}

File:SIP3.jpg

2015-06-09T17:53:02Z

John Wolfe:

{{RMImage
|Filename=File:SIP3.jpg
|CopyrightBy=
|CreationDate=2015/06/09
|Uploader=John Wolfe
|UploadDate=2015/06/09
|Description=
|Keywords=
|Format=image/jpeg
|RelatedArticles=Grandstream 410X Setup
}}

File:SIP3.jpg

2015-06-09T17:53:02Z

John Wolfe:

Grandstream 410X Setup

2015-06-09T17:30:07Z

John Wolfe:

== Grandstream GXW 410X Gateway ==

 

Description

When application of the Grandstream GXW 410X is desired for connection of FXO circuits onto the IPitomy IPBX the following configuration information can be used as a guide to install the adjunct properly.

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.

[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]

[[File:Gstream1.JPG|File:Gstream1.JPG]]

=== Procedure—Connections ===

1. Make wiring connections as necessary to the FXO circuits required.

2. Make wiring connections to the LAN per the photo above – Connect the Grandstream

WAN port to an available LAN port of the network switch/router being used on site.

[[File:Gstream2.JPG|File:Gstream2.JPG]]

=== Procedure—Power Up and Login ===

1. Power up the unit and identify its assigned IP address. (Typically assigned from the DCHP server of the host router.)

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.

3. When the Grandstream page is accessed, input the password (admin at default) and navigate to the pages below making the changes as defined.

=== IPORTANT: Procedure Line Setup ===

On first login to the Grandstream, you will be brought to the following page.

[[File:GrandstreamCPT.jpg|File:GrandstreamCPT.jpg]]

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 caue 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.

NOTE: These tests should be performed at the site and with the lines that the unit is intended to be in service upon.

=== Procedure—Configuration ===

1. Navigate to the Basic Setting page and input the data as in the example. 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.) 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. 4. Click the button. This saves information on this page before moving on. 5. Navigate to the FXO Lines page. 6. Change the Stage Method(1/2) Ch1-4:1; 7. 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 & 3. Usually only one is necessary.This data field is the routing of the calls that are received on this FXO circuit (It is reccomended to use our example on this line which is ch1:1111;ch2:2222;ch3:3333;ch4:4444; This will break out the channels individually and allow the most flexibillity in use.  We have programmed the Grandstream to route calls that have been received to the SIP Server using Profile 1.

[[File:Gstream3.JPG|File:Gstream3.JPG]]

8.Click the button. This saves information on this page before moving on.

=== Procedure—Configuration Profile ===

[[File:SIP.jpg|File:SIP.jpg]]

1. Navigate to the Profile 1 page. 2. Here the SIP Server must be programmed. Set this to be the IP Address of the IPBX. In our example, the address is 192.168.2.18. 3. Also be sure that the SIP Registration field is set to No. 4.Click the button. This saves information on this page before moving on. 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. 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; {x+ | [x*]+}

[[File:Sip2.jpg|File:Sip2.jpg]] to be able to send a * to outside IVR's 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

[[File:Gstream5.JPG|File:Gstream5.JPG]]

=== Procedure—Configuring the IPitomy IP PBX for Grandstream GXW 410x ===

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)

2. Click “Save Changes”

3. Then click on theApply Changes (upper right) to make these settings operational in the Ipitomy IPBX.

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.

 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.

=== Procedure—Configuring Optional Outbound Routing Methods ===

The Grandstream allows for two methods of outbound dialing at the same time.

Round-Robin (Linear Hunt)

 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

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.

1. Navigate to the Channels page

2. Go to the Port Scheduling Schema (Voip->PSTN) section and input the code or codes that you wish (99x)

== Remote Grandstream Configuration ==

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.

=== Grandstream Changes ===

#On the Profile page (profile 1 if that is the only one being used) set SIP Registration to YES
#Under the Channels page, configure one channel with login credentials:
#Set the SIP server in the Profile to the PBX Public IP.

[[File:Remote Device - Channels.jpg|File:Remote Device - Channels.jpg]]

=== PBX SIP Provider Changes ===

#Instead of entering an IP address for Host, set to “dynamic”
#Set Register and Authentication to YES
#Set Username and Name to the value entered as SIP User ID and Authenticate ID in the grandstream
#Set Secret to the value entered for Authen Password in the grandstream

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.

== Troubleshooting ==

=== Procedure—Troubleshooting—Outbound Dialing ===

If you having trouble dialing outbound make the following changes on the FXO Lines page:

 1. Tweak the Disconnect Threshold from 100 to 300ms.

 2. Tweak the Minimum Delay Before Dialing Out from 500 to 750ms.

=== Procedure—Troubleshooting—Call Quality ===

 If you are having call quality issues try the following changes on the Channels page:

 1.Set Silence Suppression from YES to NO.

 2. Set Echo Cancellation from YES to NO

=== Procedure—Troubleshooting—Call Volume ===

 If you are having issue with call volume try the following changes on the Channels page:

 1.Increase/Decrease Tx to PSTN Audio Gain by increments of 3 for issues with external party volume.

 2. Increase/Decrease Rx from PSTN Audio Gain by increments of 3 for issues with internal party volume

''' Procedure—Troubleshooting—Call Buzzing Noise'''

If you are having issues with a buzz heard prior to a Menu prompt; try upgrading the Grandstream firmware:

1. Navigate to Advanced Settings page

2. Ensure HTTP is selected for the method to upgrade

3. Set Firmware Server Path: to firmware.grandstream.com

4. Set Automatic Upgrade to YES

5. Set Allow DHCP Option 66 to override server to No Click Update at the bottom of the page

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

=== Procedure—Troubleshooting—Restore Factory Default ===

To Restore Factory Defaults:

1. While powered up, hold the recessed Reset button in for 7+ seconds

'''[[File:Gstream6.JPG|File:Gstream6.JPG]] '''

'''Procedure Setup of QOS'''

1. Access the IP of the Grandstream.

2. Navigate in the Grandstream Web Admin to Networks – Advanced Settings

3. Set the Layer 3 Diff Serv Value to 24 Which is CS3 or in Binary 011000

4. Submit your changes and reboot

 {{:Grandstream_FXO_FAQ}}

File:Sip2.jpg

2015-06-09T17:29:55Z

John Wolfe:

{{RMImage
|Filename=File:Sip2.jpg
|CopyrightBy=
|CreationDate=2015/06/09
|Uploader=John Wolfe
|UploadDate=2015/06/09
|Description=
|Keywords=
|Format=image/jpeg
|RelatedArticles=Grandstream 410X Setup
}}

File:Sip2.jpg

2015-06-09T17:29:55Z

John Wolfe:

File:SIP.jpg

2015-06-09T17:26:17Z

John Wolfe:

{{RMImage
|Filename=File:SIP.jpg
|CreationDate=2015/06/09
|Uploader=John Wolfe
|UploadDate=2015/06/09
|Format=image/jpeg
|RelatedArticles=Grandstream 410X Setup
}}

File:SIP.jpg

2015-06-09T17:26:17Z

John Wolfe: uploaded a new version of "File:SIP.jpg"

{{RMImage
|Filename=File:SIP.jpg
|CopyrightBy=
|CreationDate=2015/06/09
|Uploader=John Wolfe
|UploadDate=2015/06/09
|Description=
|Keywords=
|Format=image/jpeg
|RelatedArticles=Grandstream 410X Setup
}}

File:SIP.jpg

2015-06-09T17:12:17Z

John Wolfe:

File:SIP.jpg

2015-06-09T17:12:17Z

John Wolfe:

Grandstream 410X Setup

2015-05-15T18:02:49Z

John Wolfe:

== Grandstream GXW 410X Gateway ==

 

Description

When application of the Grandstream GXW 410X is desired for connection of FXO circuits onto the IPitomy IPBX the following configuration information can be used as a guide to install the adjunct properly.

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.

[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]

[[File:Gstream1.JPG|File:Gstream1.JPG]]

=== Procedure—Connections ===

1. Make wiring connections as necessary to the FXO circuits required.

2. Make wiring connections to the LAN per the photo above – Connect the Grandstream

WAN port to an available LAN port of the network switch/router being used on site.

[[File:Gstream2.JPG|File:Gstream2.JPG]]

=== Procedure—Power Up and Login ===

1. Power up the unit and identify its assigned IP address. (Typically assigned from the DCHP server of the host router.)

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.

3. When the Grandstream page is accessed, input the password (admin at default) and navigate to the pages below making the changes as defined.

=== IPORTANT: Procedure Line Setup ===

On first login to the Grandstream, you will be brought to the following page.

[[File:GrandstreamCPT.jpg|File:GrandstreamCPT.jpg]]

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 caue 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.

NOTE: These tests should be performed at the site and with the lines that the unit is intended to be in service upon.

=== Procedure—Configuration ===

1. Navigate to the ―Basic Setting page and input the data as in the example.

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.)

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.

4. Click the button. This saves information on this page before moving on.

5. Navigate to the ―FXO Lines page.

6. Change the ―Stage Method(1/2): to Ch1-4:1;

7. 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 & 3. Usually only one is necessary.This data field is the routing of the calls that are received on this FXO circuit (It is reccomended to use our example on this line which is ch1:1111;ch2:2222;ch3:3333;ch4:4444; ... This will break out the channels individually and allow the most flexibillity in use.  We have programmed the Grandstream to route calls that have been received to the SIP Server using Profile 1.

[[File:Gstream3.JPG|File:Gstream3.JPG]]

8.Click the button. This saves information on this page before moving on.

=== Procedure—Configuration Profile ===

[[File:Gstream4.jpg|File:Gstream4.jpg]]

1. Navigate to the Profile 1 page.

2. Here the SIP Server must be programmed. Set this to be the IP Address of the IPBX. In our example, the address is 192.168.2.18.

3. Also be sure that the SIP Registration field is set to ―No.

4.Click the button. This saves information on this page before moving on.

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.

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

be able to send a * to the CO. Go to Procedure Configuration Dial Plan to make these changes. Otherwise:

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

[[File:Gstream5.JPG|File:Gstream5.JPG]]

=== Procedure—Configuring the IPitomy IP PBX for Grandstream GXW 410x ===

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)

2. Click “Save Changes”

3. Then click on theApply Changes (upper right) to make these settings operational in the Ipitomy IPBX.

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.

 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.

=== Procedure—Configuring Optional Outbound Routing Methods ===

The Grandstream allows for two methods of outbound dialing at the same time.

Round-Robin (Linear Hunt)

 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

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.

1. Navigate to the Channels page

2. Go to the Port Scheduling Schema (Voip->PSTN) section and input the code or codes that you wish (99x)

== Remote Grandstream Configuration ==

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.

=== Grandstream Changes ===

#On the Profile page (profile 1 if that is the only one being used) set SIP Registration to YES
#Under the Channels page, configure one channel with login credentials:
#Set the SIP server in the Profile to the PBX Public IP.

[[File:Remote Device - Channels.jpg|File:Remote Device - Channels.jpg]]

=== PBX SIP Provider Changes ===

#Instead of entering an IP address for Host, set to “dynamic”
#Set Register and Authentication to YES
#Set Username and Name to the value entered as SIP User ID and Authenticate ID in the grandstream
#Set Secret to the value entered for Authen Password in the grandstream

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.

== Troubleshooting ==

=== Procedure—Troubleshooting—Outbound Dialing ===

If you having trouble dialing outbound make the following changes on the FXO Lines page:

 1. Tweak the Disconnect Threshold from 100 to 300ms.

 2. Tweak the Minimum Delay Before Dialing Out from 500 to 750ms.

=== Procedure—Troubleshooting—Call Quality ===

 If you are having call quality issues try the following changes on the Channels page:

 1.Set Silence Suppression from YES to NO.

 2. Set Echo Cancellation from YES to NO

=== Procedure—Troubleshooting—Call Volume ===

 If you are having issue with call volume try the following changes on the Channels page:

 1.Increase/Decrease Tx to PSTN Audio Gain by increments of 3 for issues with external party volume.

 2. Increase/Decrease Rx from PSTN Audio Gain by increments of 3 for issues with internal party volume

''' Procedure—Troubleshooting—Call Buzzing Noise'''

If you are having issues with a buzz heard prior to a Menu prompt; try upgrading the Grandstream firmware:

1. Navigate to Advanced Settings page

2. Ensure HTTP is selected for the method to upgrade

3. Set Firmware Server Path: to firmware.grandstream.com

4. Set Automatic Upgrade to YES

5. Set Allow DHCP Option 66 to override server to No Click Update at the bottom of the page

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

=== Procedure—Troubleshooting—Restore Factory Default ===

To Restore Factory Defaults:

1. While powered up, hold the recessed Reset button in for 7+ seconds

'''[[File:Gstream6.JPG|File:Gstream6.JPG]] '''

'''Procedure Setup of QOS'''

1. Access the IP of the Grandstream.

2. Navigate in the Grandstream Web Admin to Networks – Advanced Settings

3. Set the Layer 3 Diff Serv Value to 24 Which is CS3 or in Binary 011000

4. Submit your changes and reboot

 {{:Grandstream_FXO_FAQ}}

File:Gstream4.jpg

2015-05-15T18:02:39Z

John Wolfe:

{{RMImage
|Filename=File:Gstream4.jpg
|CopyrightBy=
|CreationDate=2015/05/15
|Uploader=John Wolfe
|UploadDate=2015/05/15
|Description=
|Keywords=
|Format=image/jpeg
|RelatedArticles=Grandstream 410X Setup
}}

File:Gstream4.jpg

2015-05-15T18:02:39Z

John Wolfe:

File:Gstream4.JPG

2015-05-15T17:59:26Z

John Wolfe:

{{RMImage
|Filename=File:Gstream4.JPG
|CreationDate=2013/09/12
|Uploader=John Wolfe
|UploadDate=2013/09/12
|Format=image/jpeg
|RelatedArticles=Grandstream 410X Setup
}}

File:Gstream4.JPG

2015-05-15T17:59:26Z

John Wolfe: uploaded a new version of "File:Gstream4.JPG"

{{RMImage
|Filename=File:Gstream4.JPG
|CopyrightBy=
|CreationDate=2013/09/12
|Uploader=John Wolfe
|UploadDate=2013/09/12
|Description=
|Keywords=
|Format=image/jpeg
|RelatedArticles=Grandstream 410X Setup
}}

Overhead Paging

2015-03-25T16:28:13Z

John Wolfe:

There are three possible methods to integrate overhead paging to the PBX; SIP, FXS, or FXO.

= Overhead Paging =

== Paging Devices ==

=== IPitomy SIP Paging ===

==== Using an UP220/290 as a paging port ====

Use the IPitomy IP220 HD telephone as a paging point of connection. [[File:App Note - IP220 Paging.pdf|IP220 Paging]]

==== IPitomy Overhead Speaker ====

==== IPitomy IP710 Door Box ====

Use the following guide to get the Door Box connected to your network and to any door strike.

[[File:IS710v2 Smart Sercurity IP Intercom User Manual.doc|File:IS710v2 Smart Sercurity IP Intercom User Manual.doc]]

Once on network, you can provision the Door Box as an IP2XX phone via Auto-Discovery. Log into the phone webGUI (user: root/pass: root) and program the Hotline feature so that a press of the button automatically calls whatever destination you need.

=== SIP ===

Using a SIP based overhead paging device is the best option as the device registers directly to the PBX as a SIP extension. This allows for the easiest integration as we do not need a gateway between the PBX and the paging device. In order to page, you will dial the SIP extension, and since it is registered as a SIP extension, you are also able to add the overhead to a paging ring group, along with the other extensions on the system.

We also have guides that go over the configuration of Valcom and Cyberdata SIP based paging devices.

[[File:Configuration of VIP120 Ceiling Speaker.doc|File:Configuration of VIP120 Ceiling Speaker.doc]]

[[File:Configuration of VIP172L Door phone.doc|File:Configuration of VIP172L Door phone.doc]]

[[File:Configuration of VIP201 Paging Gateway.doc|File:Configuration of VIP201 Paging Gateway.doc]]

[[File:Configuration of Cyberdata VoIP Ceiling Speaker.doc|File:Configuration of Cyberdata VoIP Ceiling Speaker.doc]]

[[File:Configuration of Cyberdata VoIP Outdoor Intercom.doc|File:Configuration of Cyberdata VoIP Outdoor Intercom.doc]]

[[File:Configuration of Cyberdata VoIP Paging Amplifier.doc|File:Configuration of Cyberdata VoIP Paging Amplifier.doc]]

=== FXS ===

An FXS paging device requires an FXS gateway ([http://www.ipitomy.com/webrelease/IPitomy/IPr20/IPr20-Quick_Guide.pdf IPr20], [http://www.ipitomy.com/webrelease/Linksys/Linksys-PAP2T-QuickGuide.pdf PAP2], [http://www.ipitomy.com/webrelease/Grandstream/Grandstream%204024%20Quick%20Guide.pdf Grandstream], or [http://wiki.ipitomy.com/index.php/Cisco_SPA112_FXS_Gateway SPA112]) to integrate as a SIP extension and is the second best option. Again, you would dial the extension to page, or you can add the extension to a ring group and page overhead along with a group of phones.

=== FXO ===

An FXO paging device requires an FXO gateway ([http://www.ipitomy.com/pbx_files/ip400/IP400-Quick_Guide.pdf IP400] or [http://www.ipitomy.com/webrelease/Grandstream/Tech%20Bulletin%202011-008%20Grandstream%20Configuration%20Guide.pdf Grandstream]) to integrate to the PBX. You then need to create a new [http://www.ipitomy.com/webrelease/IPitomy/IP1100+/Outbound_Routing_Quick_Guide.pdf Outbound Route] and add the trunk the paging device is connected to so the user can dial and access the overhead. Our experience has shown you want to be sure to strip digits in the trunk otherwise they may hear digits being dialed in the overhead.

Using an FXO paging device is the least advised method for a few reasons. The main reason is how you make the call. Instead of registering as an extension that can easily be dialed, you have to create an Outbound Route to access the gateway, meaning you cannot create a Ring Group with the paging device and some extensions included. Another reason has been our dealers experience with the FXO paging devices themselves. Sometimes special programming is needed that they are not aware of (eg. requiring an outside power source to give line voltage). It seems historically, the FXO overhead paging devices have been the most problematic so it is our advise to avoid them if possible.

== FAQ ==

=== Grandstream FXS Paging - Paging System Doesn't disconnect ===

Problem: The external paging equipment will answer and broadcast page announcement but when the user hangs up the phone the paging system doesn't disconnect.

Solution: The problem is that the Grandstream is set at default for too short a duration of loop current interruption. The paging equipment doesn't recognize the short interruption as a valid disconnect and therefore remains connected - even though the connection has been terminated between the phone and the Grandsream. The solution is to increase the loop current interruption time so that the paging equipment will detect the on-hook condition and disconnect. Grandstream sets this value at 200ms at default. We suggest changing it to 1000ms (1s) or greater.

#Navigate to the Grandstream GUI PROFILE page (usually Profile 1).
#Locate the datafield "Loop Current Disconnect" and set it to YES.
#Locate the datafield "Loop Current Disconnect Duration" and set it to: 1000.

''' '''

'''Ipitomy IP-Overhead Speaker'''

To configure the Ipitomy IP overhead Speaker once the Speaker is installed the unit can be Auto Discovered using The Auto Discovery tool. Use the following Instruction to set it up.

1) Navigate to Destinations Extensions and perform an Auto Discovery Scan of the Network. Find the MAC ID of the device in the list produced. You should also create an Extension in the PBX for the Device to use. This extension should be set as an Ipitomy 2xx Phone type. Set this as the extension in the Auto Discovery and then assign Configure and restart the Device.

[[File:1.jpg|File:1.jpg]]

Once configured you can add the speaker to any standard Page group, or more prefferably any multi cast paging group. Always remember that once added to the Multi Cast Page group the speaker must be rebooted in order to aquire the Multi Cast Page information. You can navigate into the device to verify its multi cast assignments by determining the IP and then navigating to this IP and logging in. The Log in Credentials are as any Ipitomy device with root being the user and root being the password also.

[[File:2.jpg|File:2.jpg]]

Additional: You can also determine the IP of the SIP Speaker by opening the access panel. beneath the grill of the speaker. Pressing the center button the unit will voice its IP address.

File:2.jpg

2015-03-25T16:21:06Z

John Wolfe:

{{RMImage
|Filename=File:2.jpg
|CreationDate=2015/03/06
|Uploader=John Wolfe
|UploadDate=2015/03/06
|Format=image/jpeg
|RelatedArticles=Wall Mount IP290
}}

File:2.jpg

2015-03-25T16:21:06Z

John Wolfe: uploaded a new version of "File:2.jpg"

{{RMImage
|Filename=File:2.jpg
|CopyrightBy=
|CreationDate=2015/03/06
|Uploader=John Wolfe
|UploadDate=2015/03/06
|Description=
|Keywords=
|Format=image/jpeg
|RelatedArticles=Wall Mount IP290
}}

File:1.jpg

2015-03-25T16:17:23Z

John Wolfe:

{{RMImage
|Filename=File:1.jpg
|CreationDate=2015/03/06
|Uploader=John Wolfe
|UploadDate=2015/03/06
|Format=image/jpeg
|RelatedArticles=Wall Mount IP290
}}

File:1.jpg

2015-03-25T16:17:23Z

John Wolfe: uploaded a new version of "File:1.jpg"

{{RMImage
|Filename=File:1.jpg
|CopyrightBy=
|CreationDate=2015/03/06
|Uploader=John Wolfe
|UploadDate=2015/03/06
|Description=
|Keywords=
|Format=image/jpeg
|RelatedArticles=Wall Mount IP290
}}

IS710 Door Box

2015-03-20T16:52:45Z

John Wolfe: Created page with "'''IPitomy Door Box Quick Setup Guide''' The Ipitomy IS710 Door box comes by Default configured to a Static IP of 192.168.1.153 On First login make sure Language is ..."

'''IPitomy Door Box Quick Setup Guide'''

The Ipitomy IS710 Door box comes by Default configured to a Static IP of

192.168.1.153

On First login make sure Language is set to English using the drop down on the left.

It is reccomended to connect to this IP address and then step one would be to configure for DHCP the network Settings.

[[File:One.jpg|File:One.jpg]]

Step Two Configure Extension

Simply Create an Extension for the Doorbox in the PBX. Then using Auto- Discovery find the device by scanning the network and configure in the normal Auto Discovery Method.

[[File:Two.jpg|File:Two.jpg]]

Once the Extension is registered with the PBX, the you can setup the Hotdial. This can be a Group or Extension in the system. When the button on the doorbox is pressed, then this will be the group or the extension which is called. The code to dial to send the open command to the output contacts by default is 00. This is defined under DTMF.

[[File:Three.jpg|File:Three.jpg]]

The operation is defined as, User presses Center call button on Door box, Unit Hot dials the Extension or group defined under Hotdial, and the answering party dials 00 which activates the door open current.

File:Three.jpg

2015-03-20T16:51:21Z

John Wolfe:

{{RMImage
|Filename=File:Three.jpg
|CopyrightBy=
|CreationDate=2015/03/20
|Uploader=John Wolfe
|UploadDate=2015/03/20
|Description=
|Keywords=
|Format=image/jpeg
|RelatedArticles=IS710 Door Box
}}

File:Three.jpg

2015-03-20T16:51:21Z

John Wolfe:

File:Two.jpg

2015-03-20T16:43:36Z

John Wolfe:

{{RMImage
|Filename=File:Two.jpg
|CopyrightBy=
|CreationDate=2015/03/20
|Uploader=John Wolfe
|UploadDate=2015/03/20
|Description=
|Keywords=
|Format=image/jpeg
|RelatedArticles=IS710 Door Box
}}

File:Two.jpg

2015-03-20T16:43:36Z

John Wolfe:

File:One.jpg

2015-03-20T16:12:49Z

John Wolfe:

{{RMImage
|Filename=File:One.jpg
|CopyrightBy=
|CreationDate=2015/03/20
|Uploader=John Wolfe
|UploadDate=2015/03/20
|Description=
|Keywords=
|Format=image/jpeg
|RelatedArticles=IS710 Door Box
}}