Trunk between SIP and PJSIP

Hi all, I’m new here. I hope someone can help me. I have two asterisk servers that I need to connect with a trunk, one server has only chan_sip running asterisk 13.13 and the other server only has chan_pjsip running asterisk 20.6. can someone help me with an example of what the config for the trunk should look like? Any help would be greatly appreciated.

There is no such thing as a trunk in SIP or Asterisk, but connections to another PBX are the same as connecting to an ITSP, typically with IP based matching, and you are not restricted to authenticating in just one direction (a policy feature of ITSPs, not a technical limitation).

(The word “trunk” appears nowhere in RFC 3261).

A minimalist configuration, untested, is something like:

Asterisk 13:

host=<IP of Asterisk 20>

Asterisk 20


contact=sip:<IP of Asterisk 13>

match=<IP of  Asterisk 13>

Although you would probably want to review the codecs and contexts, and consider direct media, NAT traversals, authentication, and security. I’ve assumed you are neither in North America, nor Japan

PS Please don’t use SIP to mean chan_sip. chan_pjsip is probabl a better SIP than chan_sip.

Hi David

Thanks, I’ll give it a try.

You need to try and understand what the configuration is actually doing, rather than just trying.

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