<--- 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=z9hG4bK45EDE10 CSeq: 5236 INVITE To: Content-Type: application/sdp From: ;tag=746182A9 Call-ID: 2130229940@192.168.11.123 Subject: sip:999777@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 32816 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 - 2130229940@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=z9hG4bK45EDE10;received=192.168.11.123 From: ;tag=746182A9 To: ;tag=as0665fd76 Call-ID: 2130229940@192.168.11.123 CSeq: 5236 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="14eb84fb" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2130229940@192.168.11.123' in 32000 ms (Method: INVITE) Found user '999777' <--- 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=z9hG4bK45EDE10 CSeq: 5236 ACK To: ;tag=as0665fd76 From: ;tag=746182A9 Call-ID: 2130229940@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=z9hG4bK73C42E7D CSeq: 5237 INVITE To: Proxy-Authorization: Digest username="999777", realm="asterisk", nonce="14eb84fb", uri="sip:9214020@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="00fa762d8827bcb6426e1e1e832baf65", opaque="", algorithm="MD5" Content-Type: application/sdp From: ;tag=746182A9 Call-ID: 2130229940@192.168.11.123 Subject: sip:999777@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 32816 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 - 2130229940@192.168.11.123 Found user '999777' 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 192.168.11.123:32816 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 192.168.11.123:32816 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=z9hG4bK73C42E7D;received=192.168.11.123 From: ;tag=746182A9 To: Call-ID: 2130229940@192.168.11.123 CSeq: 5237 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/999777-08208050", "SIP/214020@cisco|20") in new stack Audio is at 192.168.11.124 port 16402 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.11.120:5060: INVITE sip:214020@192.168.11.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK32660968;rport From: "999777" ;tag=as516a504f To: Contact: Call-ID: 7f96487a0167b0691131d51f422125f1@192.168.11.124 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 09 Jul 2007 11:29:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 291 v=0 o=root 19057 19057 IN IP4 192.168.11.124 s=session c=IN IP4 192.168.11.124 t=0 0 m=audio 16402 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 214020@cisco [Jul 9 15:29:44] WARNING[19133]: channel.c:3147 ast_channel_make_compatible: No path to translate from SIP/cisco-08212628(1) to SIP/999777-08208050(4) <--- SIP read from 192.168.11.120:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK32660968;rport From: "999777" ;tag=as516a504f To: ;tag=3A9AB20-1C8E Date: Mon, 09 Jul 2007 11:29:42 GMT Call-ID: 7f96487a0167b0691131d51f422125f1@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=z9hG4bK32660968;rport From: "999777" ;tag=as516a504f To: ;tag=3A9AB20-1C8E Date: Mon, 09 Jul 2007 11:29:42 GMT Call-ID: 7f96487a0167b0691131d51f422125f1@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:214020@192.168.11.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK32660968;rport From: "999777" ;tag=as516a504f To: ;tag=3A9AB20-1C8E Contact: Call-ID: 7f96487a0167b0691131d51f422125f1@192.168.11.124 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/cisco-08212628 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [9214020@local-phones:2] Congestion("SIP/999777-08208050", "") 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=z9hG4bK73C42E7D;received=192.168.11.123 From: ;tag=746182A9 To: ;tag=as4400d7a0 Call-ID: 2130229940@192.168.11.123 CSeq: 5237 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/999777-08208050' <--- 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=z9hG4bK73C42E7D CSeq: 5237 ACK To: ;tag=as4400d7a0 From: ;tag=746182A9 Call-ID: 2130229940@192.168.11.123 Content-Length: 0 User-Agent: kphone/4.2 Contact: <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '7f96487a0167b0691131d51f422125f1@192.168.11.124' Method: INVITE Really destroying SIP dialog '2130229940@192.168.11.123' Method: ACK