We have the following setup:
UPLINK <--> BRU3-CORE <--> BRU3-TRUNK <--> CUSTOMER BRU3-CORE = 11.6-cert6 with chan_sip BRU3-TRUNK = 11.6-cert4 with chan_sip CUSTOMER = 13.6.0, with pjsip
We have a SIP call coming in from UPLINK to BRU3-CORE, what gets forwarded to BRU3-TRUNK what gets forwarded to CUSTOMER.
The CUSTOMER accepts the call, and sends an 200 OK with SDP. Dtmfmode is on auto on all servers.
The 200 OK we receive from CUSTOMER is fine, however, the 200 OK that is relayed to BRU3-CORE doesn’t contain the rtpmap:101 telephone-event/8000 and fmtp:101 0-16 lines in the SDP. This gets relayed further down the chain what causes an issue that we don’t receive the DTMF events.
I assume asterisk should relay the rtpmap:101 telephone-event/8000 and fmtp:101 0-16 lines to BRU3-CORE.
SDP from CUSTOMER to BRU3-TRUNK:
v=0 o=- 249438027 249438029 IN IP4 184.108.40.206 s=Asterisk c=IN IP4 220.127.116.11 t=0 0 m=audio 16088 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv
SDP from BRU3-TRUNK to BRU3-CORE:
v=0 o=destiny 1484284222 1484284222 IN IP4 18.104.22.168 s=Destiny Voice Infrastructure c=IN IP4 22.214.171.124 t=0 0 m=audio 22566 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
Am I missing something here, or is this a known bug?