[Asterisk-Users] AAH 0.9 - SIP DTMF negotiation problem

Tue, 26 Apr 2005 14:57:02 -0700

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:303XXXXXXX[EMAIL PROTECTED]:5060;user=phone SIP/2.0
Via:SIP/2.0/UDP myprovider.net;branch=z9hG4bK.myproviderV5060-0-992912121-1986469743-1114382353904
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=1986469743-1114382353904
To: "CSI"<sip:303XXXXXXX@myprovider.net;user=phone>;tag=as2e2a4319
Call-ID: 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 - - - -
 
 
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