[asterisk-dev] How to debug reinvites not getting forwarded to other call leg (pjsip)

Floimair Florian f.floimair at commend.com
Wed Jan 12 08:16:57 CST 2022

Hi everybody!

I am currently facing an issue with SIP reINVITEs (with changed media direction) being acknowledged by Asterisk but not forwarded to the second call leg.
My setup is as follows:

Device A -> Kamailio -> Asterisk (18.9.0 chan_pjsip) -> Kamailio -> Device B

Device A sends a reinvite through Kamailio (Proxy & Registrar) to Asterisk, which answers with 200 OK.
Asterisk is configured with mohpassthrough option so any change in SDP media direction should be forwarded from A to B or vice versa.
This works in almost all cases but I do have an edge case where a linphone client sends the reinvite and Asterisk more or less
silently discards it. I tried ramping up debug level verbosity (to 5) but was unable to spot anything in regards to reinvites or any other error/mismatch as to why it is not forwarded.

So my question is: What can I do to analyze this better, maybe even add debug messages myself to the source code, but I have no clue
where the appropriate location for this would be (maybe even in libpjsip? • I’m using bundled version).

Thanks for your help in advance.

Best Regards

Florian Floimair
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20220112/e0e4b6fb/attachment.html>

More information about the asterisk-dev mailing list