I have updated Asterisk to 18 witch some fears… but it’s official, it should be ok after 1 day…
The problem is:
Calls are established, but:
For inbound calls there is no audio in any direction.
For outbound calls the voice comes in but no voice outbound.
Looks like an RTP issue?
It was working fine today until the update.
I have restarted both asterisk and nethserver, but no luck yet.
Also tried to re-apply some truck settings.
And tried to lower the RTP port range in Asterisk SIP settings from 16000-22000 to 16000-20000 (just to avoid the need to change firewall rules), but no luck yet (or is it for the internal phones only?).
It is a pjsip trunk, SIP Server Port: 5060
asterisk service:
green red TCP: 5060, 5061, 5038, 8088, 8089 UDP: 4569, 5036, 5060, 5160, 16000:22000
The ports are forwarded on a pfsense router NAT forward and firewall rules:
…edited out…
I am lost a bit about how to solve this problem, and it should be solved until tomorrow morning…
Is there an option to roll back to the previous (12? 16?) version of asterisk?
Hi @CptCharlesG,
rolling back is not tested and really tricky, I don’t think that it’s an option.
Since you are using SIP without TLS, you can debug it with sngrep. You should see where the RTP traffic is sent.
This problem could be related to sip network settings, and the update only triggered it, verify your sip and nat settings from freepbx
check asterisk -r console if you have any error
check the trunk codec, if you use open g729 you have to download the Asterisk 18 version and restart asterisk wget http://asterisk.hosting.lv/bin/codec_g729-ast180-gcc4-glibc-x86_64-pentium4.so -O /usr/lib64/asterisk/modules/codec_g729.so