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@xxx;ftag=b510c0b3970dd2d;lr=on>
From: Port 3 <sip:xxx@xxx>;tag=b510c0b3970dd2d
To: sip:xxx@xxx;tag=as5c0bc97c
Call-ID: c6dc077eaa59b3535cc42dd7a1a34f62@xx
CSeq: 786336468 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:xxx@xxx>
Content-Type: application/sdp
Content-Length: 192
Posted: Wed May 05, 2004 7:06 pm Post subject: [Asterisk-Users] Asterisk devel. - Mediatrix dtmf bug solved
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:
Quote:
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:
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@xxx;ftag=b510c0b3970dd2d;lr=on>
From: Port 3 <sip:xxx@xxx>;tag=b510c0b3970dd2d
To: sip:xxx@xxx;tag=as5c0bc97c
Call-ID: c6dc077eaa59b3535cc42dd7a1a34f62@xx
CSeq: 786336468 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:xxx@xxx>
Content-Type: application/sdp
Content-Length: 192
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