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

[Clif Jones]

Asterisk doesn't negotiate the dynamic RTP payloads so if you don't 
match the hardcoded
ones in Asterisk, the non-matching dynamic payloads don't work on the 
Asterisk side.
You should have seen Asterisk return an SDP message with:

a=rtpmap:96 telephone-event/8000

If your phone called Asterisk and offered the above.


Arek Bekiersz wrote:

>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
>
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