-- Executing StripMSD("SIP/205712896-b58b", "1") in new stack -- Executing Prefix("SIP/205712896-b58b", "31") in new stack -- Prepended prefix, new extension is 31628234048 -- Executing SetCallerID("SIP/205712896-b58b", "205712896") in new stack -- Executing Dial("SIP/205712896-b58b", "IAX2/demo-calls:d3m0@62.223.245.55/31628234048") in new stack -- Called demo-calls:d3m0@62.223.245.55/31628234048 -- Call accepted by 62.223.245.55 (format ULAW) -- Format for call is ULAW -- IAX2/62.223.245.55:4569/4 is ringing -- Hungup 'IAX2/62.223.245.55:4569/4' == Spawn extension (demo, 31628234048, 4) exited non-zero on 'SIP/205712896-b58b' sipgate*CLI> sip debug peer 205712896 SIP Debugging Enabled for IP: 62.195.51.215:5060 sipgate*CLI> Sip read: INVITE sip:0628234048@62.223.244.206 SIP/2.0 Via: SIP/2.0/UDP 62.195.51.215:5060;branch=z9hG4bK_00D0E9007BC9_T14D4B5DC Session-Expires: 1800 From: "205712896" ;tag=00D0E9007BC9_T349484507 To: Call-ID: CALL_ID69_00D0E9007BC9_T349484508@192.168.1.101 CSeq: 10092 INVITE Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,PRACK Supported: 100rel,timer User-Agent: ACT P103SLD V: 02.05 Content-Type: application/sdp Content-Length: 316 v=0 o=username 620499863 620499863 IN IP4 192.168.1.101 s=ACT P103SLD V: 02.05 c=IN IP4 62.195.51.215 t=0 0 m=audio 41000 RTP/AVP 0 18 4 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:41001 a=direction:both a=sendrecv 14 headers, 14 lines Using latest request as basis request Sending to 62.195.51.215 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 62.195.51.215:41000 Found description format PCMU Found description format G729 Found description format G723 Found description format telephone-event Capabilities: us - 0xc(ULAW|ALAW), peer - audio=0x105(G723|ULAW|G729A)/video=0x0(EMPTY), combined - 0x4(ULAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user '205712896' Looking for 0628234048 in demo list_route: hop: Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 62.195.51.215:5060;branch=z9hG4bK_00D0E9007BC9_T14D4B5DC;received=62.195.51.215;rport=5060 From: "205712896" ;tag=00D0E9007BC9_T349484507 To: ;tag=as206b7418 Call-ID: CALL_ID69_00D0E9007BC9_T349484508@192.168.1.101 CSeq: 10092 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 62.195.51.215:5060 -- Executing StripMSD("SIP/205712896-ee63", "1") in new stack -- Executing Prefix("SIP/205712896-ee63", "31") in new stack -- Prepended prefix, new extension is 31628234048 -- Executing SetCallerID("SIP/205712896-ee63", "205712896") in new stack -- Executing Dial("SIP/205712896-ee63", "IAX2/demo-calls:d3m0@62.223.245.55/31628234048") in new stack -- Called demo-calls:d3m0@62.223.245.55/31628234048 -- Call accepted by 62.223.245.55 (format ULAW) -- Format for call is ULAW We're at 62.223.244.206 port 16398 Answering with preferred capability 0x4(ULAW) Answering with preferred capability 0x8(ALAW) Transmitting (NAT): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 62.195.51.215:5060;branch=z9hG4bK_00D0E9007BC9_T14D4B5DC;received=62.195.51.215;rport=5060 From: "205712896" ;tag=00D0E9007BC9_T349484507 To: ;tag=as206b7418 Call-ID: CALL_ID69_00D0E9007BC9_T349484508@192.168.1.101 CSeq: 10092 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 188 v=0 o=root 31039 31039 IN IP4 62.223.244.206 s=session c=IN IP4 62.223.244.206 t=0 0 m=audio 16398 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - to 62.195.51.215:5060 -- IAX2/62.223.245.55:4569/1 is ringing 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:62.195.51.215 SIP/2.0 Via: SIP/2.0/UDP 62.223.244.206:5060;branch=z9hG4bK14c98ecc From: "asterisk" ;tag=as634957eb To: Contact: Call-ID: 1cf367251dcf22c841563cfc370d9b81@62.223.244.206 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Tue, 14 Sep 2004 13:53:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 62.195.51.215:5060 sipgate*CLI> Sip read: SI41000P/2.0 200 OK Via: SIP/2.0/UDP 62.223.244.206:5060;branch=z9hG4bK14c98ecc Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,PRACK Supported: 100rel,timer Accept: application/sdp From: "asterisk" ;tag=as634957eb To: Call-ID: 1cf367251dcf22c841563cfc370d9b81@62.223.244.206 CSeq: 102 OPTIONS Content-Length: 5 10 headers, 0 lines Retransmitting #1 (no NAT): OPTIONS sip:62.195.51.215 SIP/2.0 Via: SIP/2.0/UDP 62.223.244.206:5060;branch=z9hG4bK14c98ecc From: "asterisk" ;tag=as634957eb To: Contact: Call-ID: 1cf367251dcf22c841563cfc370d9b81@62.223.244.206 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Tue, 14 Sep 2004 13:53:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 to 62.195.51.215:5060 sipgate*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 62.223.244.206:5060;branch=z9hG4bK14c98ecc Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,PRACK Supported: 100rel,timer Accept: application/sdp From: "asterisk" ;tag=as634957eb To: Call-ID: 1cf367251dcf22c841563cfc370d9b81@62.223.244.206 CSeq: 102 OPTIONS Content-Length: 0 10 headers, 0 lines Sep 14 15:53:48 NOTICE[1087199936]: chan_sip.c:6551 handle_response: Peer '205712896' is now REACHABLE! Destroying call '1cf367251dcf22c841563cfc370d9b81@62.223.244.206' sipgate*CLI> Sip read: CANCEL sip:0628234048@62.223.244.206 SIP/2.0 Via: SIP/2.0/UDP 62.195.51.215:5060;branch=z9hG4bK_00D0E9007BC9_T14D4B5DC From: "205712896" ;tag=00D0E9007BC9_T349484507 To: ;tag=as206b7418 Call-ID: CALL_ID69_00D0E9007BC9_T349484508@192.168.1.101 CSeq: 10092 CANCEL Contact: Max-Forwards: 70 Content-Length: 0 9 headers, 0 lines Sending to 62.195.51.215 : 5060 (NAT) Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 62.195.51.215:5060;branch=z9hG4bK_00D0E9007BC9_T14D4B5DC;received=62.195.51.215;rport=5060 From: "205712896" ;tag=00D0E9007BC9_T349484507 To: ;tag=as206b7418 Call-ID: CALL_ID69_00D0E9007BC9_T349484508@192.168.1.101 CSeq: 10092 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 62.195.51.215:5060 -- Hungup 'IAX2/62.223.245.55:4569/1' == Spawn extension (demo, 31628234048, 4) exited non-zero on 'SIP/205712896-ee63'