Add Trunk SIP (chan_sip) Trunk

[jos50 (Gerald)]

I just install new FreePBX SNG7-PBX16-64bit-2302-1
I notice that in Add Trunk the +Add SIP (chan_sip) Trunk is missing.
I need SIP (chan_sip) Trunk for Freephoneline
What do I do?

Thanks

[comtech]

Enable Chan SIP on Freepbx 16

Not recommended, but if you need to absolutely must enable it, you can do so in Settings, Advanced Settings: [image] You will need to restart asterisk after it’s configured.

[Stewart1 (Stewart)]

Back in 2018, pjsip was not compatible with trunks hosted by Fibernetics, including FPL.

The problem was that pjsip sends the 200 OK response to an incoming INVITE with a Contact header that has no username. This is perfectly valid according to the RFCs, but it somehow caused the Fibernetics server to not respond with an ACK, so incoming calls failed.

There was a long thread about this, in which I suggested various ways of getting it fixed, but there were no further updates.

Migrating sip to pjsip trunk problem, incoming call drops after 32 seconds

Hello, I am trying to migrate one of my sip trunks to pjsip, with no success. Everything works, except incoming calls are dropped after 32 seconds. While everything points to NAT problem, I can not figure why this is happening and which pjsip configuration file has to be changed. I am using FreePBX 14 and asterisk 13. When call comes on standard sip trunk, INVITE is sent from provider, and replied with 100 trying followed by 200 OK. Afterwards, ACK is sent from provider. In pjsip case, ACK is …

AFAIK, pjsip still sends 200 OK with no username in Contact, though it’s possible that the behavior of Fibernetics servers has changed.

If not, it will take some effort to get this fixed and you should probably use chan_sip, at least in the short term.