<--- SIP read from 192.168.11.123:5060 ---> INVITE sip:9722490@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK3F982AF1 CSeq: 6021 INVITE To: Content-Type: application/sdp From: ;tag=2B2117DC Call-ID: 1246610069@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 32840 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 - 1246610069@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=z9hG4bK3F982AF1;received=192.168.11.123 From: ;tag=2B2117DC To: ;tag=as1b0c06c3 Call-ID: 1246610069@192.168.11.123 CSeq: 6021 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56209a8e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1246610069@192.168.11.123' in 32000 ms (Method: INVITE) Found user '214059' <--- SIP read from 192.168.11.123:5060 ---> ACK sip:9722490@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK3F982AF1 CSeq: 6021 ACK To: ;tag=as1b0c06c3 From: ;tag=2B2117DC Call-ID: 1246610069@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:9722490@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK4D37957B CSeq: 6022 INVITE To: Proxy-Authorization: Digest username="214059", realm="asterisk", nonce="56209a8e", uri="sip:9722490@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="2b5c67b4ccb947e1a79c0408f2035eae", opaque="", algorithm="MD5" Content-Type: application/sdp From: ;tag=2B2117DC Call-ID: 1246610069@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 32840 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 - 1246610069@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 192.168.11.123:32840 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 192.168.11.123:32840 Looking for 9722490 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=z9hG4bK4D37957B;received=192.168.11.123 From: ;tag=2B2117DC To: Call-ID: 1246610069@192.168.11.123 CSeq: 6022 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [9722490@local-phones:1] Dial("SIP/214059-08208090", "SIP/722490@cisco|20") in new stack Audio is at 192.168.11.124 port 13740 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:722490@192.168.11.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK6692ff3f;rport From: "214059" ;tag=as4571c879 To: Contact: Call-ID: 5f77a2a00e63b7cf64d1d7c4137a7bbd@192.168.11.124 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 09 Jul 2007 11:41:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 291 v=0 o=root 19181 19181 IN IP4 192.168.11.124 s=session c=IN IP4 192.168.11.124 t=0 0 m=audio 13740 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 722490@cisco [Jul 9 15:41:55] WARNING[19237]: channel.c:3147 ast_channel_make_compatible: No path to translate from SIP/cisco-082127a0(1) to SIP/214059-08208090(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=z9hG4bK6692ff3f;rport From: "214059" ;tag=as4571c879 To: ;tag=3B4D441-6D3 Date: Mon, 09 Jul 2007 11:41:53 GMT Call-ID: 5f77a2a00e63b7cf64d1d7c4137a7bbd@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=z9hG4bK6692ff3f;rport From: "214059" ;tag=as4571c879 To: ;tag=3B4D441-6D3 Date: Mon, 09 Jul 2007 11:41:53 GMT Call-ID: 5f77a2a00e63b7cf64d1d7c4137a7bbd@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:722490@192.168.11.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK6692ff3f;rport From: "214059" ;tag=as4571c879 To: ;tag=3B4D441-6D3 Contact: Call-ID: 5f77a2a00e63b7cf64d1d7c4137a7bbd@192.168.11.124 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/cisco-082127a0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [9722490@local-phones:2] Congestion("SIP/214059-08208090", "") 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=z9hG4bK4D37957B;received=192.168.11.123 From: ;tag=2B2117DC To: ;tag=as42a1bf30 Call-ID: 1246610069@192.168.11.123 CSeq: 6022 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, 9722490, 2) exited non-zero on 'SIP/214059-08208090' <--- SIP read from 192.168.11.123:5060 ---> ACK sip:9722490@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK4D37957B CSeq: 6022 ACK To: ;tag=as42a1bf30 From: ;tag=2B2117DC Call-ID: 1246610069@192.168.11.123 Content-Length: 0 User-Agent: kphone/4.2 Contact: <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '5f77a2a00e63b7cf64d1d7c4137a7bbd@192.168.11.124' Method: INVITE Really destroying SIP dialog '1246610069@192.168.11.123' Method: ACK