<--- SIP read from 192.168.11.123:5060 ---> INVITE sip:9214020@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK2AB2B9A9 CSeq: 3692 INVITE To: Content-Type: application/sdp From: ;tag=7C04A406 Call-ID: 1039122264@192.168.11.123 Subject: sip:214059@192.168.11.124 Content-Length: 232 User-Agent: kphone/4.2 Contact: v=0 o=username 0 0 IN IP4 192.168.11.123 s=The Funky Flow c=IN IP4 192.168.11.123 t=0 0 m=audio 32848 RTP/AVP 0 97 8 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 <-------------> --- (11 headers 11 lines) --- Sending to 192.168.11.123 : 5060 (no NAT) Using INVITE request as basis request - 1039122264@192.168.11.123 <--- Reliably Transmitting (no NAT) to 192.168.11.123:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK2AB2B9A9;received=192.168.11.123 From: ;tag=7C04A406 To: ;tag=as3e16b601 Call-ID: 1039122264@192.168.11.123 CSeq: 3692 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26959360" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1039122264@192.168.11.123' in 32000 ms (Method: INVITE) Found user '214059' <--- SIP read from 192.168.11.123:5060 ---> ACK sip:9214020@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK2AB2B9A9 CSeq: 3692 ACK To: ;tag=as3e16b601 From: ;tag=7C04A406 Call-ID: 1039122264@192.168.11.123 Content-Length: 0 User-Agent: kphone/4.2 Contact: <-------------> --- (9 headers 0 lines) --- <--- SIP read from 192.168.11.123:5060 ---> INVITE sip:9214020@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK4C589DF4 CSeq: 3693 INVITE To: Proxy-Authorization: Digest username="214059", realm="asterisk", nonce="26959360", uri="sip:9214020@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="14699d4868b687d1d6c2cc55a81a8f61", opaque="", algorithm="MD5" Content-Type: application/sdp From: ;tag=7C04A406 Call-ID: 1039122264@192.168.11.123 Subject: sip:214059@192.168.11.124 Content-Length: 232 User-Agent: kphone/4.2 Contact: v=0 o=username 0 0 IN IP4 192.168.11.123 s=The Funky Flow c=IN IP4 192.168.11.123 t=0 0 m=audio 32848 RTP/AVP 0 97 8 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 <-------------> --- (12 headers 11 lines) --- Sending to 192.168.11.123 : 5060 (no NAT) Using INVITE request as basis request - 1039122264@192.168.11.123 Found user '214059' Found RTP audio format 0 Found RTP audio format 97 Found RTP audio format 8 Found RTP audio format 3 Peer audio RTP is at port 192.168.11.123:32848 Found description format PCMU for ID 0 Found description format GSM for ID 3 Found description format PCMA for ID 8 Found description format iLBC for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.11.123:32848 Looking for 9214020 in local-phones (domain 192.168.11.124) list_route: hop: <--- Transmitting (no NAT) to 192.168.11.123:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK4C589DF4;received=192.168.11.123 From: ;tag=7C04A406 To: Call-ID: 1039122264@192.168.11.123 CSeq: 3693 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [9214020@local-phones:1] Dial("SIP/214059-0820c130", "SIP/9214020@192.168.11.120|20") in new stack Audio is at 192.168.11.124 port 10086 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726aal2) to SDP Adding codec 0x20 (adpcm) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x80 (lpc10) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x800 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.11.120:5060: INVITE sip:9214020@192.168.11.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK71b2fd18;rport From: "214059" ;tag=as4b314b04 To: Contact: Call-ID: 3a02b0ff44a1d55e42c7b5c45030e8d6@192.168.11.124 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 05 Jul 2007 07:58:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 473 v=0 o=root 4528 4528 IN IP4 192.168.11.124 s=session c=IN IP4 192.168.11.124 t=0 0 m=audio 10086 RTP/AVP 0 3 8 112 5 10 7 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 9214020@192.168.11.120 <--- SIP read from 192.168.11.120:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK71b2fd18;rport From: "214059" ;tag=as4b314b04 To: ;tag=4C5F5BC5-BBC Date: Thu, 05 Jul 2007 07:58:04 GMT Call-ID: 3a02b0ff44a1d55e42c7b5c45030e8d6@192.168.11.124 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 192.168.11.120:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK71b2fd18;rport From: "214059" ;tag=as4b314b04 To: ;tag=4C5F5BC5-BBC Date: Thu, 05 Jul 2007 07:58:04 GMT Call-ID: 3a02b0ff44a1d55e42c7b5c45030e8d6@192.168.11.124 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 192.168.11.120:5060: ACK sip:9214020@192.168.11.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK71b2fd18;rport From: "214059" ;tag=as4b314b04 To: ;tag=4C5F5BC5-BBC Contact: Call-ID: 3a02b0ff44a1d55e42c7b5c45030e8d6@192.168.11.124 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/192.168.11.120-08211040 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [9214020@local-phones:2] Congestion("SIP/214059-0820c130", "") in new stack <--- Transmitting (no NAT) to 192.168.11.123:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK4C589DF4;received=192.168.11.123 From: ;tag=7C04A406 To: ;tag=as4fb7fa69 Call-ID: 1039122264@192.168.11.123 CSeq: 3693 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 <------------> == Spawn extension (local-phones, 9214020, 2) exited non-zero on 'SIP/214059-0820c130' <--- SIP read from 192.168.11.123:5060 ---> ACK sip:9214020@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK4C589DF4 CSeq: 3693 ACK To: ;tag=as4fb7fa69 From: ;tag=7C04A406 Call-ID: 1039122264@192.168.11.123 Content-Length: 0 User-Agent: kphone/4.2 Contact: <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '3a02b0ff44a1d55e42c7b5c45030e8d6@192.168.11.124' Method: INVITE Really destroying SIP dialog '1039122264@192.168.11.123' Method: ACK