Marty Mastera
2005-Apr-26 13:23 UTC
[Asterisk-Users] AAH 0.9 - SIP DTMF negotiation problem
I'm having a problem with SIP dtmf negotiation during call setup. My provider wants me to use rfc2833, which I configured in the general section of sip.conf but it's not working. From packet capture and sip debug we see that my provider is offering 0 and 105 (0=ulaw, 105=a codec used on their cisco boxes). When AAH responds to the invite with 200 OK, it's agreeing on 0 (ulaw), but not specifiying/requesting/etc.. 101 for rfc2833 dtmf. Is it by design that AAH 0.9 (asterisk 1.0.7) won't request rtp payload type 101 unless the invite offered it? I'm told that even if the invite didn't offer it, I should expect asterisk to send a 200 OK asking for rfc2833 since I specified it in sip.conf... On a side note, once an incoming call is up, if I do a show channel on that call asterisk says that it's using rfc2833, but since it never asked for that in the 200 OK, the far end isn't using it (hence why I'm not getting any inbound dtmf I assume)...My provider has tested against his switch with a cvs version of asterisk, same scenario where his invite doesn't offer rfc2833, but in his case the cvs asterisk sends a 200 OK with 101 specfied and things work like they should. I tried looking at the code to see if I could force 101 to be sent in the ok, but I'm having a hard time figuring out where that would go...any help in that area would be appreciated. Marty Sip read: INVITE sip:303XXXXXXX1@10.0.2.3:5060;user=phone SIP/2.0 Via:SIP/2.0/UDP myprovider.net;branch=z9hG4bK.myproviderV5060-0-992912121-1986469743-111 4382353904 From:"COLORADO CALL "<sip:303XXXXXXX@myprovider.net;user=phone>;tag=1986469743-1114382353904 To:"CSI"<sip:303XXXXXXX@myprovider.net;user=phone> Call-ID:183913904240405-862404306@myprovider.net CSeq:992912121 INVITE Contact:<sip:myprovider.net:5060> Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY Supported:100rel Accept:application/sdp,application/dtmf Max-Forwards:10 Content-Type:application/sdp Content-Length:311 v=0 o=BroadWorks 111617 1 IN IP4 192.168.0.3 s=- c=IN IP4 192.168.0.3 t=0 0 m=audio 18164 RTP/AVP 0 105 a=rtpmap:105 X-NSE/8000 a=fmtp:105 192-194,200-202 a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 105 a=X-cpar: a=rtpmap:105 X-NSE/8000 a=X-cpar: a=fmtp:105 192-194,200-202 a=X-cap: 2 image udptl t38 13 headers, 13 lines Using latest request as basis request Sending to 192.168.0.3 : 5060 (non-NAT) Found peer myprovider.net' Found RTP audio format 0 Found RTP audio format 105 Peer audio RTP is at port 192.168.0.3:18164 Found description format X-NSE Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 3039960401 in custom-sip-did list_route: hop: <sip:myprovider.net:5060> SIP/2.0 200 OK Via: SIP/2.0/UDP myprovider.net;branch=z9hG4bK.myprovider.netV5060-0-992912121-1986469743 -1114382353904 From: "COLORADO CALL"<sip:303XXXXXXX@myprovider.net;user=phone>;tag=1986469743-111438235 3904 To: "CSI"<sip:303XXXXXXX@myprovider.net;user=phone>;tag=as2e2a4319 Call-ID: 183913904240405-862404306@myprovider.net <mailto:183913904240405-862404306@myprovider.net> CSeq: 992912121 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:303XXXXXXX@10.0.2.3> Content-Type: application/sdp Content-Length: 150 v=0 o=root 1076 1076 IN IP4 10.0.2.3 s=session c=IN IP4 10.0.2.3 t=0 0 m=audio 14384 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050426/9dd61a68/attachment.htm