Hello!
I believe I am having a codec mismatch issue causing calls to terminate immediately after 200 OK. I think this is happening because of a supplied optional encoding parameter in the SDP (tools.ietf.org/html/rfc4566#section-5 page 25,26). I’ve included the text below of the SDP from the original INVITE and the SDP from my 200 OK. Please note the a=rtpmap:0 PCMU/8000/1
The INVITE is from a ZTE switch. I am running Freepbx 13.0.192.18 with asterisk 13.17.0. Can anyone confirm if this is the issue and, even better, have a solution that I can implement with FreePBX? Many thanks!
ZTE INVITE SDP:
v=0
o=- 565458263 565458263 IN IP4 192.168.0.100
s=-
c=IN IP4 192.168.0.100
t=0 0
m=audio 50144 RTP/AVP 0 97 98 12 99 100 96
c=IN IP4 192.168.0.100
a=rtpmap:0 PCMU/8000/1
a=rtpmap:97 EVRC/8000/1
a=rtpmap:98 EVRC0/8000/1
a=rtpmap:12 QCELP/8000/1
a=rtpmap:99 QCELP8K/8000/1
a=rtpmap:100 IWF/8000/1
a=rtpmap:96 telephone-event/8000/1
a=fmtp:96 0-15
a=sendrecv
FreePBX 200 OK SDP:
v=0
o=- 1070621862 1070621862 IN IP4 172.16.0.1
s= Asterisk
c=IN IP4 172.16.0.1
t=0 0
m=audio 15280 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv