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<DIV><SPAN class=895045119-26042005><FONT face=Arial size=2>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.</FONT></SPAN></DIV>
<DIV><SPAN class=895045119-26042005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=895045119-26042005><FONT face=Arial size=2>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...</FONT></SPAN></DIV>
<DIV><SPAN class=895045119-26042005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=895045119-26042005><FONT face=Arial size=2>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.</FONT></SPAN></DIV>
<DIV><SPAN class=895045119-26042005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=895045119-26042005><FONT face=Arial size=2>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.</FONT></SPAN></DIV>
<DIV><SPAN class=895045119-26042005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=895045119-26042005>Marty</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=895045119-26042005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=895045119-26042005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Sip read:<BR>INVITE sip:303<SPAN
class=895045119-26042005>XXXXXXX</SPAN>1@10.0.2.3:5060;user=phone
SIP/2.0<BR>Via:SIP/2.0/UDP <SPAN
class=895045119-26042005>myprovider.net</SPAN>;branch=z9hG4bK.<SPAN
class=895045119-26042005>myprovider</SPAN>V5060-0-992912121-1986469743-1114382353904<BR>From:"<SPAN
class=895045119-26042005>COLORADO CALL</SPAN> "<sip:303<SPAN
class=895045119-26042005>XXXXXXX</SPAN>@<SPAN
class=895045119-26042005>myprovider.net</SPAN>;user=phone>;tag=1986469743-1114382353904<BR>To:"<SPAN
class=895045119-26042005>CSI</SPAN>"<sip:303<SPAN
class=895045119-26042005>XXXXXXX</SPAN>@<SPAN
class=895045119-26042005>myprovider.net</SPAN>;user=phone><BR>Call-ID:183913904240405-862404306@<SPAN
class=895045119-26042005>myprovider</SPAN>.net<BR>CSeq:992912121
INVITE<BR>Contact:<sip:<SPAN
class=895045119-26042005>myprovider</SPAN>.net:5060><BR>Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY<BR>Supported:100rel<BR>Accept:application/sdp,application/dtmf<BR>Max-Forwards:10<BR>Content-Type:application/sdp<BR>Content-Length:311</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>v=0<BR>o=BroadWorks 111617 1 IN IP4 192.168.<SPAN
class=895045119-26042005>0</SPAN>.<SPAN
class=895045119-26042005>3</SPAN><BR>s=-<BR>c=IN IP4 192.168.<SPAN
class=895045119-26042005>0.3</SPAN><BR>t=0 0<BR><STRONG><FONT size=3>m=audio
18164 RTP/AVP 0 105</FONT></STRONG><BR>a=rtpmap:105 X-NSE/8000<BR>a=fmtp:105
192-194,200-202<BR>a=X-sqn:0<BR>a=X-cap: 1 audio RTP/AVP 105<BR>a=X-cpar:
a=rtpmap:105 X-NSE/8000<BR>a=X-cpar: a=fmtp:105 192-194,200-202<BR>a=X-cap: 2
image udptl t38</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>13 headers, 13 lines<BR>Using latest request as
basis request<BR>Sending to 192.168.<SPAN
class=895045119-26042005>0</SPAN>.<SPAN class=895045119-26042005>3</SPAN> : 5060
(non-NAT)<BR>Found peer <SPAN
class=895045119-26042005>myprovider</SPAN>.net'<BR><STRONG><FONT size=3>Found
RTP audio format 0<BR></FONT></STRONG>Found RTP audio format 105<BR>Peer audio
RTP is at port 192.168.<SPAN class=895045119-26042005>0</SPAN>.<SPAN
class=895045119-26042005>3</SPAN>:18164<BR>Found description format
X-NSE<BR><STRONG><FONT size=3>Capabilities: us - 0x4 (ulaw), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)<BR></FONT></STRONG>Non-codec
capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0
(nothing)<BR>Looking for 3039960401 in custom-sip-did<BR>list_route: hop:
<sip:<SPAN
class=895045119-26042005>myprovider</SPAN>.net:5060></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP <SPAN
class=895045119-26042005>myprovider</SPAN>.net;branch=z9hG4bK.<SPAN
class=895045119-26042005>myprovider</SPAN>.netV5060-0-992912121-1986469743-1114382353904<BR>From:
"<SPAN class=895045119-26042005>COLORADO CALL</SPAN>"<sip:303<SPAN
class=895045119-26042005>XXXXXXX</SPAN>@<SPAN
class=895045119-26042005>myprovider</SPAN>.net;user=phone>;tag=1986469743-1114382353904<BR>To:
"<SPAN class=895045119-26042005>CSI</SPAN>"<sip:303<SPAN
class=895045119-26042005>XXXXXXX</SPAN>@<SPAN
class=895045119-26042005>myprovider</SPAN>.net;user=phone>;tag=as2e2a4319<BR>Call-ID:
</FONT><A href="mailto:183913904240405-862404306@myprovider.net"><FONT
face=Arial size=2>183913904240405-862404306@<SPAN
class=895045119-26042005>myprovider</SPAN>.net</FONT></A><BR><FONT face=Arial
size=2>CSeq: 992912121 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER<BR>Contact: <sip:303<SPAN
class=895045119-26042005>XXXXXXX</SPAN>@10.0.2.3><BR>Content-Type:
application/sdp<BR>Content-Length: 150</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>v=0<BR>o=root 1076 1076 IN IP4
10.0.2.3<BR>s=session<BR>c=IN IP4 10.0.2.3<BR>t=0 0<BR><STRONG><FONT
size=3>m=audio 14384 RTP/AVP 0<BR></FONT></STRONG>a=rtpmap:0
PCMU/8000<BR>a=silenceSupp:off - - - -</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=895045119-26042005></SPAN></FONT> </DIV></BODY></HTML>