chan_sip

chan_sip is the original SIP channel driver from Asterisk that many have grown to love or hate over the years. Replaced by chan_pjsip in Asterisk 12, it was deprecated in Asterisk 16, disabled from building by default in Asterisk 19, and was removed from the Asterisk master branch in commit 4095a382da630f8af2aac2d223fcaa70c03b81ba on January 3, 2023 (in anticipation of removal for Asterisk 21, to be released in late 2023).

If you are new to Asterisk or getting started with SIP, you should almost certainly use chan_pjsip, a newer, more standards-compliant, more feature-rich SIP channel driver. chan_pjsip (based on pjproject's PJSIP stack) is actively core supported by Sangoma.

chan_sip is maintained here for the benefit of those who still need or prefer it, but you are advised to avoid using it if possible. This version of chan_sip is primarily maintained by a community Asterisk developer, and is not official or endorsed by Sangoma in any way. Please do not contact Sangoma or the Asterisk project with issues - any issues should be filed in this repo only.

Usage

Manual

This assumes you are familiar with compiling Asterisk from source. If not, see below.

• Navigate into your Asterisk source directory, either before or after compiling

• Run wget https://raw.githubusercontent.com/InterLinked1/chan_sip/master/chan_sip_reinclude.sh && chmod +x chan_sip_reinclude.sh && ./chan_sip_reinclude.sh

• Run menuselect/menuselect --enable chan_sip menuselect.makeopts to choose to build chan_sip, since chan_sip does not build by default.

• Run make channels && make install to finish compiling and installing Asterisk with these changes

• Don't forget to load => chan_sip in modules.conf, since chan_sip is disabled by default.

Automatic

You can use PhreakScript to fully compile and install an enhanced version of Asterisk automatically, including with chan_sip. Simply specify the -s or --sip options to the phreaknet install command and chan_sip will be automatically installed, e.g. run phreaknet install -s. Note that if if you install from master or version 21+, it will pull in the enhanced chan_sip from this repo. If you install using version 20 (current default) or older, it will simply build the included version of chan_sip from those versions (which contain some of the enhancements included below, but not all of them).

Don't forget to load => chan_sip in modules.conf, since chan_sip is disabled by default.

About This Repo

This is not a fork of Asterisk itself with chan_sip still in it. It is, in fact, the opposite: a differential fork of chan_sip from current Asterisk so that chan_sip can easily continue to be used with modern Asterisk systems. This allows users to upgrade to versions of Asterisk that no longer have chan_sip officially, but still have it. The goal is to provide those who require chan_sip the ability to use it on modern systems.

This version of chan_sip was forked from chan_sip immediately prior to the commit removing it. However, it has since diverged and includes a number of improvements that were not merged into Asterisk previously or have been developed since then. These changes are stated below to make it clear what has changed.

The chan_sip unit tests are from the Asterisk test suite prior to removal: https://gerrit.asterisk.org/c/testsuite/+/19612

Changes

• Support for custom parameters (SIPAddParameter application; SIP_PARAMETER function; send_oli, uri_parameters_instead sip.conf config options). This functionality is now also available in chan_pjsip. This is included in PhreakScript for all versions. Initial patch: https://github.com/InterLinked1/phreakscript/blob/master/patches/sipcustparams.patch and abandoned code review: https://gerrit.asterisk.org/c/asterisk/+/16569

• Support for fax control. This functionality has also been available in chan_pjsip. This is included in PhreakScript for all versions. Initial patch: https://github.com/InterLinked1/phreakscript/blob/master/patches/sipfaxcontrol.diff

• Support for Homer (HEP) in chan_sip. This functionality has also been available in chan_pjsip.

• Support for macros has been removed per the Asterisk deprecation policy

All this said, as both an Asterisk user and developer myself, I recommend you migrate to chan_pjsip if you don't have a good reason not to. It will likely serve your needs better. If, however, you still need or prefer chan_sip, then this repo is for you.

Development

I do not actively add functionality to chan_sip myself - however, occasionally I do patches for it on request, and I open source these when possible. Any such patches will be incorporated here, in the event they are useful to anyone. All contributions are released under GPLv2 (the same license that Asterisk has).

Improvements or new features are accepted to this repo. However, having learned some things from chan_sip's existence in the official Asterisk tree, please follow these guidelines:

• This repo uses the Asterisk Coding Guidelines, Asterisk Commit Guidelines, and Asterisk CHANGES documentation

• The directory structure of this repo mirrors the directory structure of the Asterisk repo.

• Minimize changes to existing files to the extent possible, especially chan_sip.c, which remains a huge, monolithic source file. New functionality should be added in separate files when possible: both source and header files.

• Write tests when possible using the Asterisk Test Suite.

To-Do List

• The usecallmanager patches are not currently incorporated here because they present a huge merge conflict.