[Asterisk-Users] Asterisk devel. - Mediatrix dtmf bug solved

[Arek Bekiersz]

Hello,


When using Asterisk version 0.7.2, FreeBSD port with Mediatrix 1124 gateway,
there is problem with DTMF "out-of-band".

See debug below: Mediatrix forces (*) to use Payload Type as 96:

[...]
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
[...]

Then we've got this nice debug from (*):
May  5 10:48:15 NOTICE[139648000]: rtp.c:418 ast_rtp_read: Unknown RTP codec
96 received
May  5 10:48:15 NOTICE[139648000]: rtp.c:418 ast_rtp_read: Unknown RTP codec
96 received

We had this static_RTP_PT[xx] structure in rtp.c :(asterisk source):
[...]
[34] = {1, AST_FORMAT_H263},
[97] = {1, AST_FORMAT_ILBC},
[101] = {0, AST_RTP_DTMF},
[110] = {1, AST_FORMAT_SPEEX},
[121] = {0, AST_RTP_CISCO_DTMF}, // Must be type 121
[...]

as there is no 96 entry and function ast_rtp_read() is returning 'Unknown
RTP.....".

We added entry, recompiled Asterisk and yeah it works!!!!!!
See debug below:

[...]
Answering with preferred capability 8
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx;branch=z9hG4bK0167.9f4aff84.0
Via: SIP/2.0/UDP xxx;branch=z9hG4bK6b3a5b06f
Record-Route: <sip:xxx at xxx;ftag=b510c0b3970dd2d;lr=on>
From: Port 3 <sip:xxx at xxx>;tag=b510c0b3970dd2d
To: sip:xxx at xxx;tag=as5c0bc97c
Call-ID: c6dc077eaa59b3535cc42dd7a1a34f62 at xx
CSeq: 786336468 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:xxx at xxx>
Content-Type: application/sdp
Content-Length: 192

v=0
o=root 62170 62170 IN IP4 xxxx
s=session
c=IN IP4 xxx
t=0 0
m=audio 13784 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
to xxx:5060



I will ask Mediatrix what they think about it.

Regards,
Arek Bekiersz

arek at perceval.net
Perceval R&D Team