I setup a Pbxact uc40 and it’s fully update on 16.0.30 and 12.7.8-2208-2.sng7. I had setup 2 remote phones, a Sangoma S505 provisioned with EPM and a Snom M100 base with 2 M109 cordless phones. When I originally set them up, they worked great. I tried to test them before shipping them out to the end user, and now I have no audio either way. The phone rings then the call drops after 30 seconds due to no rtp stream. Even a *43 echo test has no audio. The public ip of the remote phones are trusted in the phone system.
With RTP debug on I see this:
[2023-01-24 21:20:23] ERROR[26448][C-0000002d]: pbx_functions.c:651 ast_func_read2: Function SIP_HEADER not registered
It had worked without any port forwards on the remote phone router, but to troubleshoot I also created a port forward in the remote router for ports 5060-5160 udp and tcp and udp ports 10000-20000.
I also switched from asterisk 18 to 20 with no change.
Seems like it has to be something simple but I haven’t been able to figure it out yet!
As you seem to be in the Show Me State, show me the SIP trace. At the Asterisk command prompt, type pjsip set logger on
make another failing call and paste the Asterisk log (not the console log) for the call.
This has been down for me since at least yesterday. If the server isn’t down, I suppose it is possible that it has been blacklisted because of illegal content (most of the posting are, at best, spam).
The immediate problem is that Asterisk is putting its private address 192.168.11.2 in both Contact header and SDP when responding to ext. 201. It should be providing its public address in those fields.
However, I don’t know what setting is responsible.
In Asterisk SIP settings, confirm that Local Networks and External Address are correctly set.
On the chan_pjsip tab, External IP Address and Local Network for each transport should be left blank.
If you change any of these settings, after Submit and Apply Config you must restart Asterisk.
If the above is inapplicable or doesn’t help, please post the contents of /etc/asterisk/pjsip.transports.conf
as well as pjsip.transports_custom.conf and/or pjsip.transports_custom_post.conf, if present.
Also paste a new log, if you made any changes above.
In General SIP settings, Local Networks and External address are correct. On chan_pjsip external ip and local network are blank.
Under transports, here are the settings.
Wow, I’m very puzzled. Just a guess, confirm that for the extensions in question, RTP Symmetric, Rewrite Contact and Force rport are all Yes, and try setting Direct Media to No.
If no luck, possible the WebRTC stuff is somehow interfering. In Asterisk SIP Settings, try turning off any STUN or TURN settings. (This will break WebRTC, which you need, but neither of the extensions involved in your test use it, so the test should be able to complete without it.)
Direct Media was set to yes, I had enabled earlier in troubleshooting.
I tested it again before disabling it, and now I have audio!
I had changed the sip settings> External ip address from dynamic dns name to the automatically detect setting and under chan pjsip Certificate manager changed the certificate from non to the LE one.
