IPv6 configuration for Enswitch - Revision history

Rodolfojcj: Brief note to ensure that firewall rules do not block IPv6 network traffic

2025-11-24T21:21:00Z

Brief note to ensure that firewall rules do not block IPv6 network traffic

← Older revision		Revision as of 21:21, 24 November 2025
Line 8:		Line 8:

	* The IPv6 addresses to use are mutually reachable for both the server and the client sides. Normally those IPv6 addresses will be public ones. Using local IPv6 addresses will not work.		* The IPv6 addresses to use are mutually reachable for both the server and the client sides. Normally those IPv6 addresses will be public ones. Using local IPv6 addresses will not work.

			* Ensure that no firewall rules block IPv6 traffic between the server and client sides. You should be able to either ping the involved IPv6 addresses or see IPv6 packets arriving on the server. If needed, capture network traffic with tools like tcpdump or sngrep and then inspect it with Wireshark to check that expected packets are transmitted.

	* The IPv6 addresses to use are already configured, enabled and working in the operating system where the Enswitch related services are running.		* The IPv6 addresses to use are already configured, enabled and working in the operating system where the Enswitch related services are running.

Rodolfojcj: Typo fix

2024-08-23T23:23:43Z

Typo fix

← Older revision		Revision as of 23:23, 23 August 2024
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	Most of the installations of Enswitch use only IPv4 addresses for all the networking traffic, however working with IPv6 addresses for telephony services is also possible, which needs some adjustments that are explained ~~bellow~~.		Most of the installations of Enswitch use only IPv4 addresses for all the networking traffic, however working with IPv6 addresses for telephony services is also possible, which needs some adjustments that are explained below.

	== Basic considerations ==		== Basic considerations ==
Line 41:		Line 41:
	* Be sure that the operating system where Asterisk runs has a properly configured IPv6 address, according to the initial considerations.		* Be sure that the operating system where Asterisk runs has a properly configured IPv6 address, according to the initial considerations.

	* Either the traditional ''SIP'' stack or the newer ''PJSIP'' project can be used, then choose the steps ~~bellow~~ according to your requirements.		* Either the traditional ''SIP'' stack or the newer ''PJSIP'' project can be used, then choose the steps below according to your requirements.

	=== If using the traditional Asterisk ''SIP'' stack ===		=== If using the traditional Asterisk ''SIP'' stack ===

Rodolfojcj: Added note for the Enswitch setting related with the SIP stack used with Asterisk

2022-01-05T17:45:24Z

Added note for the Enswitch setting related with the SIP stack used with Asterisk

← Older revision		Revision as of 17:45, 5 January 2022
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	* If using Enswitch 4.2 or earlier, go to ''Global > Configuration'' and change the '''''Use sipsak command for monitoring Asterisk machines''''' setting to '''''No'''''.		* If using Enswitch 4.2 or earlier, go to ''Global > Configuration'' and change the '''''Use sipsak command for monitoring Asterisk machines''''' setting to '''''No'''''.

			* If using Enswitch 4.2 or newer and the ''PJSIP'' stack for Asterisk, go to ''Global > Configuration'' and change the '''''Default SIP protocol driver''''' setting to '''''PJSIP'''''.

	* For the corresponding customer, add the telephone line(s) that will make and receive the calls for the created IPv6 domain.		* For the corresponding customer, add the telephone line(s) that will make and receive the calls for the created IPv6 domain.

Rodolfojcj: Added details for using the PJSIP stack and IPv6 with Asterisk

2022-01-05T17:39:42Z

Added details for using the PJSIP stack and IPv6 with Asterisk

@@ Line 40: / Line 40: @@
 * Be sure that the operating system where Asterisk runs has a properly configured IPv6 address, according to the initial considerations.
 * To make Asterisk listen on IPv6 addresses, edit the file ''/etc/asterisk/sip.conf'', so that its ''[general]'' section looks like this example:
@@ Line 76: / Line 80: @@
   t38pt_udptl = yes
 * Restart Asterisk, with a command like this:
@@ Line 136: / Line 188: @@
 |Ask your system administrator the correct details for the IPv6 address to use and configure them in the operating system.
 |-
-|Asterisk does not process calls.
+|Asterisk does not process calls when using the traditional ''SIP'' stack.
 |Asterisk ''sip.conf'' configuration file is not ready to handle IPv6 traffic. To check this, the command '''''netstat -anptu <nowiki>|</nowiki> grep -i asterisk''''' could be used, which will show something like '''''udp        0      0 0.0.0.0:5070            0.0.0.0:*                           958684/asterisk''''' if only IPv4 traffic is working, while something like '''''udp6       0      0 :::5070                 :::*                                988099/asterisk''''' will appear if both IPv4 and IPv6 traffic are working.
 |Edit that ''sip.conf'' file as explained before.
 |-
 |Kamailio does not accept calls, or connection attempts fail from the client to port 5060 on the IPv6 Kamailio address.
@@ Line 146: / Line 202: @@
 |The SIP registration seems to work, but calls work intermittently, or are not authorized, or the server does not process the ''INVITEs'' requests or keeps asking for registration, or simply the calls can not be established at all.
 |The network communication latency between the client and the server is too high.
-|Get the client and server networks to communicate with the less posible latency. As a reference, take the following measured ''ping'' times from the client to the server in one of the tested scenarios:
+|Get the client and server networks to communicate with the less possible latency. As a reference, take the following measured ''ping'' times from the client to the server in one of the tested scenarios:
 * With an average response time of 73 milliseconds, the calls did not work.
 * With an average response time of 12 milliseconds, the calls worked fine.

Rodolfojcj: Minor typo fix

2021-08-13T00:39:51Z

Minor typo fix

← Older revision		Revision as of 00:39, 13 August 2021
Line 109:		Line 109:
	== Client related changes ==		== Client related changes ==

	The client could be a softphone or a physical phone. The important consideration is to properly ~~configured~~ it with an IPv6 address that can communicate with the ''Kamailio'' and ''Asterisk'' servers that are on the Enswitch server side.		The client could be a softphone or a physical phone. The important consideration is to properly configure it with an IPv6 address that can communicate with the ''Kamailio'' and ''Asterisk'' servers that are on the Enswitch server side.

	== How to make calls? ==		== How to make calls? ==

Rodolfojcj: Configuration of Enswitch and related services to work with IPv6 addresses

2021-08-07T20:27:58Z

Configuration of Enswitch and related services to work with IPv6 addresses

New page

Most of the installations of Enswitch use only IPv4 addresses for all the networking traffic, however working with IPv6 addresses for telephony services is also possible, which needs some adjustments that are explained bellow.

== Basic considerations ==

* Enswitch 3.15 or newer is used.

* Enswitch sources are updated to their most recent revision.

* The IPv6 addresses to use are mutually reachable for both the server and the client sides. Normally those IPv6 addresses will be public ones. Using local IPv6 addresses will not work.

* The IPv6 addresses to use are already configured, enabled and working in the operating system where the Enswitch related services are running.

* The details in this page are based on a single machine deployment that is used to run Enswitch, Kamailio and Asterisk services.

* The sample public IPv6 address ''2001:41d0:302:2200::20b8'' will be used to illustrate the needed configurations. In case we need to quickly enable it in the operating system, these commands could be used:

ip addr add 2001:41d0:302:2200::20b8 dev ens3
ip -6 route add 2001:41d0:302:2200::1 dev ens3
ip -6 route add default via 2001:41d0:302:2200::1 dev ens3

== Kamailio ==

* Be sure that the operating system where Kamailio runs has a properly configured IPv6 address, according to the previous considerations.

* In the file ''/etc/kamailio/kamailio.cfg'' add a line like the following one, adjusted for your IPv6 address:

listen=[2001:41d0:302:2200::20b8]:5060

Possibly, there could be other ''listen'' lines in that file, for example for an IPv4 address.

* Optionally, increase the debug level in that same file ''/etc/kamailio/kamailio.cfg'', by having a line like this one:

debug = 2

* Restart Kamailio, with a command like this:

systemctl restart kamailio.service

== Asterisk ==

* Be sure that the operating system where Asterisk runs has a properly configured IPv6 address, according to the initial considerations.

* To make Asterisk listen on IPv6 addresses, edit the file ''/etc/asterisk/sip.conf'', so that its ''[general]'' section looks like this example:

[general]
...
; the initial bindaddr and bindport options are commented out
;bindaddr = 0.0.0.0
;bindport = 5070
; and a new line is added to bind to IPv6 addresses
bindaddr=[::]:5070
...

* Again edit the file ''/etc/asterisk/sip.conf'' to configure the sections that correspond to the Kamailio IPv6 address, like this example:

; this is the first one
<nowiki>[[2001:41d0:302:2200::20b8]]</nowiki>
host = [2001:41d0:302:2200::20b8]
fromdomain = [2001:41d0:302:2200::20b8]
type = friend
insecure = port,invite
context = from-internal
canreinvite = no
nat = yes
t38pt_udptl = yes

; this is the second one, which is similar but adds the Kamailio port to the address
<nowiki>[[2001:41d0:302:2200::20b8]:5060]</nowiki>
host = [2001:41d0:302:2200::20b8]
fromdomain = [2001:41d0:302:2200::20b8]
type = friend
insecure = port,invite
context = from-internal
canreinvite = no
nat = yes
t38pt_udptl = yes

As a remainder, the assumption is that both Kamailio and Asterisk run on the same machine and use the same IPv6 address, but if that is not the case for you, then adjust those sections with the IPv6 address of your Kamailio server.

* Restart Asterisk, with a command like this:

systemctl restart asterisk.service

== Enswitch settings ==

In the web UI, do the following:

* For each IPv6 address that will be used as a valid destination that will accept calls, go to ''Global > Domain'' to add a new domain, entering that IPv6 address as the domain name. The value to enter is a valid representation of the IPv6 address (short format or long format), without the surrounding square brackes. As an example, any of the following equivalent addresses will be valid:

2001:41d0:302:2200::20b8
2001:41d0:0302:2200:0000:0000:0000:20b8

* Go to ''Global > Machines'' to add a new Asterisk machine, entering the IPv6 address as the value for the machine IP address field, following the same guidelines and examples given above.

* If using Enswitch 4.2 or earlier, go to ''Global > Configuration'' and change the '''''Use sipsak command for monitoring Asterisk machines''''' setting to '''''No'''''.

* For the corresponding customer, add the telephone line(s) that will make and receive the calls for the created IPv6 domain.

* For the corresponding customer, add any telephony feature that will also use the created IPv6 domain. An example of this is a ''feature code''.

== Enswitch services ==

Restart the Enswitch services with the ''enswitch restart'' command, or alternatively with these commands:

systemctl restart enswitch_messaged
systemctl restart enswitch_routed
systemctl restart enswitch_sipd

== Client related changes ==

The client could be a softphone or a physical phone. The important consideration is to properly configured it with an IPv6 address that can communicate with the ''Kamailio'' and ''Asterisk'' servers that are on the Enswitch server side.

== How to make calls? ==

Let's suppose that phone line with name ''3478082'' wants to call feature code ''200'' that is configured for the IPv6 domain with address ''2001:41d0:302:2200::20b8''. For this case, these things will happen:

* Phone line ''3478082'' will register as ''<nowiki>sip:3478082@[2001:41d0:302:2200::20b8]</nowiki>'' with the server.
* Phone line ''3478082'' will dial to destination ''<nowiki>sip:200@[2001:41d0:302:2200::20b8]</nowiki>''.
* The call processing will happen on the server side similarly to a call that uses IPv4 addresses, with Enswitch services, Kamailio and Asterisk communicating with each other using IPv6 traffic as needed.

== In case of failures ==

The following is a summary of the failures and errors observed while developing and testing Enswitch to work with IPv6 address.

{| class="wikitable"
! Problem
! Possible cause
! Possible fix
|-
|Client receives the error ''SIP registration failed, status=403 (URI domain not configured locally, relaying forbidden)''.
|The domain has not been added to the system, then it is not allowed.
|In the Enswitch web UI, add the new domain as explained before.
|-
|There is no IPv6 connectivity to the server side.
|The server IPv6 address is local and is not available outside the server operating system. If the command '''''ip address show''''' is run, its output for a local IPv6 address will show something like '''''inet6 fe80::f816:3eff:fe94:93c1/64 scope link''''', while a public IPv6 address will show something like '''''inet6 2001:41d0:302:2200::20b8/128 scope global'''''.
|Ask your system administrator the correct details for the IPv6 address to use and configure them in the operating system.
|-
|Asterisk does not process calls.
|Asterisk ''sip.conf'' configuration file is not ready to handle IPv6 traffic. To check this, the command '''''netstat -anptu <nowiki>|</nowiki> grep -i asterisk''''' could be used, which will show something like '''''udp 0 0 0.0.0.0:5070 0.0.0.0:* 958684/asterisk''''' if only IPv4 traffic is working, while something like '''''udp6 0 0 :::5070 :::* 988099/asterisk''''' will appear if both IPv4 and IPv6 traffic are working.
|Edit that ''sip.conf'' file as explained before.
|-
|Kamailio does not accept calls, or connection attempts fail from the client to port 5060 on the IPv6 Kamailio address.
|Kamailio ''/etc/kamailio/kamailio.cfg'' configuration file is not ready to handle IPv6 traffic. To check this, the command '''''netstat -anptu <nowiki>|</nowiki> grep -i kamailio''''' could be used, which will show something like '''''udp6 0 0 2001:41d0:302:2200:5060 :::* 943729/kamailio''''' if IPv6 traffic is working.
|Edit that ''/etc/kamailio/kamailio.cfg'' file as explained before.
|-
|The SIP registration seems to work, but calls work intermittently, or are not authorized, or the server does not process the ''INVITEs'' requests or keeps asking for registration, or simply the calls can not be established at all.
|The network communication latency between the client and the server is too high.
|Get the client and server networks to communicate with the less posible latency. As a reference, take the following measured ''ping'' times from the client to the server in one of the tested scenarios:
* With an average response time of 73 milliseconds, the calls did not work.
* With an average response time of 12 milliseconds, the calls worked fine.
|-
|The client receives error messages saying ''SIP registration failed, status=403 (Not allowed)''.
|The required configuration is either incomplete, incorrect or missing.
|Check each of the previous sections to be sure that each of the related services is properly configured.
|-
|}