Arek Bekiersz
2004-May-05 08:06 UTC
[Asterisk-Users] Asterisk devel. - Mediatrix dtmf bug solved
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@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
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@perceval.net
Perceval R&D Team
Clif Jones
2004-May-05 12:06 UTC
[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:>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@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 > >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@perceval.net >Perceval R&D Team > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >