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

[Marty Mastera]

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 at 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 at myprovider.net;user=phone>;tag=1986469743-1114382353904
To:"CSI"<sip:303XXXXXXX at myprovider.net;user=phone>
Call-ID:183913904240405-862404306 at 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 at myprovider.net;user=phone>;tag=1986469743-111438235
3904
To: "CSI"<sip:303XXXXXXX at myprovider.net;user=phone>;tag=as2e2a4319
Call-ID: 183913904240405-862404306 at myprovider.net
<mailto:183913904240405-862404306 at myprovider.net> 
CSeq: 992912121 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:303XXXXXXX at 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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