[asterisk-dev] DTMF problem when receiving call

[Maxim Samo]

Hi,

First time I post to asterisk-dev and I'm not really a developer. I do
believe though that I have come across a bug but would be happy to stand
corrected.

I have a conference bridge that calls me with g.722 codec to join a
conference. When I receive the call from the bridge I have to press "1"
to be transferred to the conference. The DTMF though never gets
recognized by the bridge.

Analyzing the traffic and looking at the SDP packets I see that:

- The conference bridge sends an INVITE but without SDP. This to remain
as flexible as possible.
- My asterisk (I'm on 1.6.1.10) sends SDP as follows:

v=0
o=root 446369530 446369530 IN IP4 1.2.3.4
s=Asterisk PBX 1.6.1.10
c=IN IP4 1.2.3.4
b=CT:384
t=0 0
m=audio 13584 RTP/AVP 9 0 8 18 3
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 11062 RTP/AVP 31 34 98
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=sendrecv

- note that you cannot see a telephone-event in there.
- Any DTMF that I then send will be sent as rfc2833 however since I
never advertized that in SDP the other side won't accept them.

When I initiate the call the SDP looks like this:

v=0
o=root 788318273 788318273 IN IP4 212.147.15.36
s=Asterisk PBX 1.6.1.10
c=IN IP4 212.147.15.36
t=0 0
m=audio 14042 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Note the telephone-event.

In sip.conf I have dtmfmode=rfc2833

Any clues to what's going on here?

best regards,

Maxim Samo