Router Info - Revision history

Mike Lunn: /* Sophos */

2026-04-08T20:25:12Z

Sophos

Mike Lunn: /* Fortigate #2 */

2022-11-30T18:24:46Z

Fortigate #2

Mike Lunn: /* Ensure Outbound Traffic is Allowed from PBX */

2022-09-30T19:12:08Z

Ensure Outbound Traffic is Allowed from PBX

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Mike Lunn: /* Fortigate */

2021-11-18T19:58:29Z

Fortigate

Mike Lunn: /* Fortigate */

2021-11-18T19:57:37Z

Fortigate

Mike Lunn at 14:05, 19 August 2019

2019-08-19T14:05:53Z

Mike Lunn: /* Comcast Modem */

2014-09-19T19:33:37Z

Comcast Modem

Mike Lunn at 20:06, 16 September 2014

2014-09-16T20:06:09Z

Mike Lunn at 18:27, 28 July 2014

2014-07-28T18:27:55Z

Mike Lunn at 19:37, 11 June 2014

2014-06-11T19:37:02Z

← Older revision		Revision as of 20:25, 8 April 2026
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	Some Sophos models have a hidden SIP module that is not in any way indicated, nor accessible, from the webgui. It must be disabled from the command line console. If left enabled, it attempts to override any rules you may have in place for sip/rtp traffic and can result in one-way audio, and other issues with calls successfully connecting.		Some Sophos models have a hidden SIP module that is not in any way indicated, nor accessible, from the webgui. It must be disabled from the command line console. If left enabled, it attempts to override any rules you may have in place for sip/rtp traffic and can result in one-way audio, and other issues with calls successfully connecting.
		+
		+	== TP-link ER605 ==
		+	[[File:TPlink.jpg\|border\|left\|thumb]]
		+	Apparently you need to have h323 alg turned on for voip to work on the old Ipitomy phones.

← Older revision		Revision as of 18:24, 30 November 2022
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	end		end
		+
		+	== Fortigate #3 ==
		+
		+
		+	'''For Fortigate Firewalls running OSv7 using IPitomy Cloud PBX'''
		+
		+	After testing the Fortigate series firewalls and working with Fortigate support, we have come up with these modifications to Fortigate Firewalls to ensure they work well with IPitomy Cloud PBX’s.
		+
		+	'''If the firewall is not correctly configured, users with these devices will run into the following issues using a Fortigate:'''
		+
		+	1. Dropped calls
		+
		+	2. One way or no way audio
		+
		+	3. Poor Voice Quality/Clicking
		+
		+	4. Potential device registration issues
		+
		+	5. Duplicate SIP Ports and port shuffling which result in poor quality voice.
		+
		+	To mitigate some of these issues, '''Strict Register''' should be ''disabled (outlined below)'' to stop all phones from using a pinhole through port 65476 (external) and 5060 (internal), aka ALG management. That may be all you need to do with some PBX’s, especially premise. These instructions are designed for CLOUD IPitomy PBX’s.
		+
		+	'''''NOTE:''''' With FortiOS version 7 comes a change to profiles. It used to be that there was a default Security Policy for each Security Policy, and you could change it. With version 7 you cannot change the default policy, you must make a clone or use the premade clone called “Base”.
		+
		+	'''''ADDITIONALLY!''''' When editing Firewall policies, to add security profiles, the policy Inspection Mode MUST be set to Proxy-Based, otherwise you don’t see any Security policy except Default.
		+

	= Cisco Pix 506/501/515 and Cisco ASA<br/> =		= Cisco Pix 506/501/515 and Cisco ASA<br/> =

@@ Line 80: / Line 80: @@
 If you see this behavior, disable the NAT switch on the IPv4 policy.<br/>
-[[File:fortigatenat.PNG]]
+[[File:fortigatenat.png]]
 ==Ensure Outbound Traffic is Allowed from PBX==

@@ Line 17: / Line 17: @@
 = Fortigate<br/> =
-I found this online about solving issues with Fortigate routers and NO AUDIO with remote SIP:
+==Disable SIP ALG==
-In the command line of the fortigate type the following:
+====Check the ID of the SIP session helper====
-*<code>config system settings</code>
+  config system session-helper
-*<code>set sip-helper disable</code>
+      show
-Reboot the device
+Among the displayed settings will be one similar to the following example:
-In the command line type the following:
+ edit 13
-*<code>config system session-helper</code>
+Here entry 13 is the one which points to SIP traffic which uses UDP port 5060 for signaling.
-(now look for SIP, mostly it will be "12")
+In this example, the next commands to remove the corresponding entry would be:
-*<code>delete 12</code>
+ delete 13
-Don't use any protection profiles on the firewall of the sip rules.
+Note that it is not necessary for the SIP entry to be 13, so cross verify which entry has the sip helper settings.
 = Cisco Pix 506/501/515 and Cisco ASA<br/> =

← Older revision		Revision as of 14:05, 19 August 2019
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	We have received some information from our dealers that if your site has a Comcast modem/router, you should request a SMC and not a Linksys, as the reports are that the SMC handles VoIP more consistently. Additionally, there may be issues with Comcast modem/routers ability to handle multiple concurrent NAT sessions, limiting the number of remote phones you can install at a remote site.		We have received some information from our dealers that if your site has a Comcast modem/router, you should request a SMC and not a Linksys, as the reports are that the SMC handles VoIP more consistently. Additionally, there may be issues with Comcast modem/routers ability to handle multiple concurrent NAT sessions, limiting the number of remote phones you can install at a remote site.
		+
		+	= Sophos =
		+
		+	Some Sophos models have a hidden SIP module that is not in any way indicated, nor accessible, from the webgui. It must be disabled from the command line console. If left enabled, it attempts to override any rules you may have in place for sip/rtp traffic and can result in one-way audio, and other issues with calls successfully connecting.

← Older revision		Revision as of 19:33, 19 September 2014
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	= Comcast Modem =		= Comcast Modem =

−	We have received some information from our dealers that if your site has a Comcast modem/router, you should request a SMC and not a Linksys, as the reports are that the SMC handles VoIP more consistently.	+	We have received some information from our dealers that if your site has a Comcast modem/router, you should request a SMC and not a Linksys, as the reports are that the SMC handles VoIP more consistently. Additionally, there may be issues with Comcast modem/routers ability to handle multiple concurrent NAT sessions, limiting the number of remote phones you can install at a remote site.

← Older revision		Revision as of 20:06, 16 September 2014
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	http://www.dslreports.com/faq/verizonfios/3.0_Networking#16077		http://www.dslreports.com/faq/verizonfios/3.0_Networking#16077
		+
		+	= Comcast Modem =
		+
		+	We have received some information from our dealers that if your site has a Comcast modem/router, you should request a SMC and not a Linksys, as the reports are that the SMC handles VoIP more consistently.

← Older revision		Revision as of 18:27, 28 July 2014
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	<br/>[[File:Peplink Config.pdf\|File:Peplink Config.pdf]]		<br/>[[File:Peplink Config.pdf\|File:Peplink Config.pdf]]
		+
		+	= FIOS ActionTec =
		+
		+	We have found the following article that outlines some possible configurations that are available on the Actiontec Modem/Router combo that FIOS is installing. This gives some options on ways to configure to optimize VoIP and SIP traffic passing to remote.
		+
		+	http://www.dslreports.com/faq/verizonfios/3.0_Networking#16077

@@ Line 1: / Line 1: @@
 This page contains general information about port forwarding and disabling application layer gateways on particular routers.
@@ Line 5: / Line 4: @@
 = Sonicwall =
 {{:Router Info Sonicwall}}
@@ Line 46: / Line 46: @@
 #no fixup protocol sip udp 5060
-=Adtran=
+= Adtran =
-From a recent interaction with an AdTran tech, it was shown to us there is a setting for "proxy transparency" that needs to be enabled in order for all of the SIP traffic to pass unhindered.  This was when the Adtran was the routing device at the remote site, but likely would need to be enabled when the Adtran is at the PBX site.  Its worth trying for sure.
+From a recent interaction with an AdTran tech, it was shown to us there is a setting for "proxy transparency" that needs to be enabled in order for all of the SIP traffic to pass unhindered. This was when the Adtran was the routing device at the remote site, but likely would need to be enabled when the Adtran is at the PBX site. Its worth trying for sure.
-=PepLink=
+= PepLink =
 Here is a document sent to a dealer from PepLink regarding configuration settings that may be required for Remote SIP to function properly: