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<DIV><FONT face=Arial size=2>Hello all,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>taking examples from various pointers, I am
attempting to put together an outbound dialing example using SIP (Cisco 7960)
with 2 X100P. Everything seems to be working without generating errors,
but the problem is the phone hangs up (102/Bye). Any
pointers/advice are much appreciated</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Here is the section in
extensions.conf:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>extensions.conf</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>; From CISCO at work<BR>;<BR>exten =>
_9NXXXXXX,1,StripMSD,1<BR>exten =>
_NXXXXXX,2,Dial(Zap/2/BYEXTENSION)</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>And the debug output:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV> </DIV><FONT face=Arial size=2>
<DIV><BR>Sip read:<BR>INVITE sip:94732771@yyy.yy.113.56;user=phone
SIP/2.0<BR>Via: SIP/2.0/UDP xxx.xxx.226.166:5060<BR>From: "TOMS"
<sip:TOMS@yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c<BR>To:
<sip:94732771@yyy.yy.113.56;user=phone><BR>Call-ID: <A
href="mailto:0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166">0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166</A><BR>CSeq:
101 INVITE<BR>User-Agent: CSCO/4<BR>Contact:
<sip:TOMS@xxx.xxx.226.166:5060><BR>Expires: 180<BR>Content-Type:
application/sdp<BR>Content-Length: 250<BR>Accept: application/sdp</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=Cisco-SIPUA 20902 8155 IN IP4 xxx.xxx.226.166<BR>s=SIP
Call<BR>c=IN IP4 xxx.xxx.226.166<BR>t=0 0<BR>m=audio 27632 RTP/AVP 0 8 18
101<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:18
G729/8000<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 0-15</DIV>
<DIV> </DIV>
<DIV>12 headers, 11 lines<BR>Using latest request as basis request<BR>Sending to
xxx.xxx.226.166 : 5060 (non-NAT)<BR>Found audio format UNKN<BR>Found audio
format ALAW<BR>Found audio format UNKN<BR>Found audio format UNKN<BR>Found
description format PCMU<BR>Found description format PCMA<BR>Found description
format G729<BR>Found description format telephone-event<BR>Capabilities: us -
2147483647, them - 268/0, combined - 268<BR>Non-codec capabilities: us - 1, them
- 1, combined - 1<BR>Looking for 94732771 in from-sip<BR>list_route: hop:
<sip:TOMS@xxx.xxx.226.166:5060><BR>Transmitting (NAT):<BR>SIP/2.0 100
Trying<BR>Via: SIP/2.0/UDP
xxx.xxx.226.166:5060;received=xxx.xxx.226.166<BR>From: "TOMS"
<sip:TOMS@yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c<BR>To:
<sip:94732771@yyy.yy.113.56;user=phone>;tag=as4bfbf62f<BR>Call-ID: <A
href="mailto:0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166">0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166</A><BR>CSeq:
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:94732771@yyy.yy.113.56><BR>Content-Length:
0</DIV>
<DIV> </DIV>
<DIV><BR> to xxx.xxx.226.166:5060<BR>We're at yyy.yy.113.56 port
11818<BR>Answering with preferred capability 2147483647<BR>Answering with
non-codec capability 1<BR>Reliably Transmitting (NAT):<BR>SIP/2.0 200 OK<BR>Via:
SIP/2.0/UDP xxx.xxx.226.166:5060;received=xxx.xxx.226.166<BR>From: "TOMS"
<sip:TOMS@yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c<BR>To:
<sip:94732771@yyy.yy.113.56;user=phone>;tag=as4bfbf62f<BR>Call-ID: <A
href="mailto:0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166">0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166</A><BR>CSeq:
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:94732771@yyy.yy.113.56><BR>Content-Type:
application/sdp<BR>Content-Length: 167</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=root 25762 25762 IN IP4 yyy.yy.113.56<BR>s=session<BR>c=IN IP4
yyy.yy.113.56<BR>t=0 0<BR>m=audio 11818 RTP/AVP 101<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101 0-16</DIV>
<DIV> </DIV>
<DIV> to xxx.xxx.226.166:5060<BR>Sip read:<BR>ACK
sip:94732771@yyy.yy.113.56:5060 SIP/2.0<BR>Via: SIP/2.0/UDP
xxx.xxx.226.166:5060<BR>From: "TOMS"
<sip:TOMS@yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c<BR>To:
<sip:94732771@yyy.yy.113.56;user=phone>;tag=as4bfbf62f<BR>Call-ID: <A
href="mailto:0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166">0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166</A><BR>CSeq:
101 ACK<BR>User-Agent: CSCO/4<BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR>8 headers, 0 lines<BR>Sip read:<BR>BYE sip:94732771@yyy.yy.113.56:5060
SIP/2.0<BR>Via: SIP/2.0/UDP xxx.xxx.226.166:5060<BR>From: "TOMS"
<sip:TOMS@yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c<BR>To:
<sip:94732771@yyy.yy.113.56;user=phone>;tag=as4bfbf62f<BR>Call-ID: <A
href="mailto:0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166">0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166</A><BR>CSeq:
102 BYE<BR>User-Agent: CSCO/4<BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR>8 headers, 0 lines<BR>Sending to xxx.xxx.226.166 : 5060
(NAT)<BR>Transmitting (NAT):<BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
xxx.xxx.226.166:5060;received=xxx.xxx.226.166<BR>From: "TOMS"
<sip:TOMS@yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c<BR>To:
<sip:94732771@yyy.yy.113.56;user=phone>;tag=as4bfbf62f<BR>Call-ID: <A
href="mailto:0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166">0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166</A><BR>CSeq:
102 BYE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER<BR>Contact: <sip:94732771@yyy.yy.113.56><BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR> to xxx.xxx.226.166:5060<BR>Expression is
'1'</FONT></DIV></BODY></HTML>