PJSIP softphone not working - while SIP does - is PJSIP unsupported for SIP softphone?

First of all,

Thank you jami-developers:
I love jami! :heart: Thank you so much for that great Open Source solution!!!
After a week of trying to fix a WebRTC scenario in FOP2 you saved my day(s)! + I am sure there will be lots of future use cases to me where to use jami. Great!!!

Issue with PJSIP vs SIP:
Today I installed jami on Ubuntu 20.10. I did use the apt approach. Snap was stuck…

My goal:
Use jami (PJ)SIP softphone to send and receive phone calls via FreePBX v14 and v15 asterisk system.

SIP working fine:
SIP works out of the box.

PJSIP does not connect at all:
PJSIP on “dring” seems to not exist. Connections to PJSIP Extensions fail.
Error on Asterisk: chan_sip.c: Registration from ‘“SIP” sip:2000@192.168.0.199’ failed for ‘172.16.1.229:5061’ - Wrong password

So jami softphone only uses SIP right?
Could you at least add a warning info box to jami softphone settings that PJSIP is unsupported?
A lot of people like mine will try that…

Curious:
I have read a lot of the contribution from the jami developers towards PJSIP and vice versa.

Is PJSIP really unsupported on jami softphone?

We do run multiple PJSIP extensions it would be a pity to have to revert them back to SIP just to use jami softphone.

Kind regards and thanks once more for that great peace of software,
Raphael

snap install --edge jami should work now, Amin is working to move it to the stable snap channel

I think you are talking about https://wiki.asterisk.org/wiki/display/AST/Channel+Driver+Modules ?

I don’t know what changes for Asterisk with this driver, but Jami uses pjsip for a lot of parts. Every Jami calls is a SIP calls, so a lot of code we use is for a modified version of pjsip (cause we add some features that are not upstreamed yet).

But Jami is a SIP client or a Jami client. If I read this: https://support.flowroute.com/086810-Chan_SIP-and-Chan_PJSIP it allows asterisk to use some WebRTC extension. And indeed, Jami do not support webrtc.

However, I am not sure to understand your full point:

Error on Asterisk: chan_sip.c: Registration from ‘“SIP” sip:2000@192.168.0.199’ failed for ‘172.16.1.229:5061’ - Wrong password

it seems to use chan_sip not chan_pjsip.

PJSIP on “dring” seems to not exist. Connections to PJSIP Extensions fail.

Is this line from Asterisk? I don’t really know what are these extensions, but it may be easy to add? Imho this will need investigations to know what is different between a SIP account and the pjsip extensions, but yeah I don’t think we support this kind of account