[Asterisk-Users] SIP to PSTN gateway

[Rasta Man]

Hello all,   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   Here is the section in extensions.conf:   extensions.conf   ; From CISCO at work ; exten => _9NXXXXXX,1,StripMSD,1 exten => _NXXXXXX,2,Dial(Zap/2/BYEXTENSION)     And the debug output:     Sip read: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.226.166:5060 From: "TOMS" <sip:[EMAIL PROTECTED]>;tag=0003e362fc0a001a407788de-1f443a6c To: <sip:[EMAIL PROTECTED];user=phone> Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: CSCO/4 Contact: <sip:[EMAIL PROTECTED]:5060> Expires: 180 Content-Type: application/sdp Content-Length: 250 Accept: application/sdp   v=0 o=Cisco-SIPUA 20902 8155 IN IP4 xxx.xxx.226.166 s=SIP Call c=IN IP4 xxx.xxx.226.166 t=0 0 m=audio 27632 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15   12 headers, 11 lines Using latest request as basis request Sending to xxx.xxx.226.166 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 2147483647, them - 268/0, combined - 268 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 94732771 in from-sip list_route: hop: <sip:[EMAIL PROTECTED]:5060> Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.226.166:5060;received=xxx.xxx.226.166 From: "TOMS" <sip:[EMAIL PROTECTED]>;tag=0003e362fc0a001a407788de-1f443a6c To: <sip:[EMAIL PROTECTED];user=phone>;tag=as4bfbf62f Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0    to xxx.xxx.226.166:5060 We're at yyy.yy.113.56 port 11818 Answering with preferred capability 2147483647 Answering with non-codec capability 1 Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.226.166:5060;received=xxx.xxx.226.166 From: "TOMS" <sip:[EMAIL PROTECTED]>;tag=0003e362fc0a001a407788de-1f443a6c To: <sip:[EMAIL PROTECTED];user=phone>;tag=as4bfbf62f Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 167   v=0 o=root 25762 25762 IN IP4 yyy.yy.113.56 s=session c=IN IP4 yyy.yy.113.56 t=0 0 m=audio 11818 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16    to xxx.xxx.226.166:5060 Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.226.166:5060 From: "TOMS" <sip:[EMAIL PROTECTED]>;tag=0003e362fc0a001a407788de-1f443a6c To: <sip:[EMAIL PROTECTED];user=phone>;tag=as4bfbf62f Call-ID: [EMAIL PROTECTED] CSeq: 101 ACK User-Agent: CSCO/4 Content-Length: 0   8 headers, 0 lines Sip read: BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.226.166:5060 From: "TOMS" <sip:[EMAIL PROTECTED]>;tag=0003e362fc0a001a407788de-1f443a6c To: <sip:[EMAIL PROTECTED];user=phone>;tag=as4bfbf62f Call-ID: [EMAIL PROTECTED] CSeq: 102 BYE User-Agent: CSCO/4 Content-Length: 0   8 headers, 0 lines Sending to xxx.xxx.226.166 : 5060 (NAT) Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.226.166:5060;received=xxx.xxx.226.166 From: "TOMS" <sip:[EMAIL PROTECTED]>;tag=0003e362fc0a001a407788de-1f443a6c To: <sip:[EMAIL PROTECTED];user=phone>;tag=as4bfbf62f Call-ID: [EMAIL PROTECTED] CSeq: 102 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0    to xxx.xxx.226.166:5060 _expression_ is '1'