<--- SIP read from 10.4.20.184:5060 ---> INVITE sip:1000@10.4.20.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK223040 To: From: "Joe S. Phigan" ;tag=7531 Call-ID: 1180028884-3040-TOPLAP@10.4.20.184 CSeq: 897 INVITE Max-Forwards: 20 User-Agent: Uplink Skype2Sip 1.1 Contact: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 284 v=0 o=- 836899966 836899988 IN IP4 10.4.20.184 s=Uplink Skype2Sip c=IN IP4 10.4.20.184 t=0 0 m=audio 10000 RTP/AVP 0 8 3 13 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:13 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (13 headers 14 lines) --- Sending to 10.4.20.184 : 5060 (NAT) Using INVITE request as basis request - 1180028884-3040-TOPLAP@10.4.20.184 <--- Reliably Transmitting (NAT) to 10.4.20.184:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK223040;received=10.4.20.184;rport=5060 From: "Joe S. Phigan" ;tag=7531 To: ;tag=as79f6c802 Call-ID: 1180028884-3040-TOPLAP@10.4.20.184 CSeq: 897 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0163fef3" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1180028884-3040-TOPLAP@10.4.20.184' in 32000 ms (Method: INVITE) Found user '7101' Itchy*CLI> <--- SIP read from 10.4.20.184:5060 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- Itchy*CLI> <--- SIP read from 10.4.20.184:5060 ---> ACK sip:1000@10.4.20.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK223040 To: ;tag=as79f6c802 From: "Joe S. Phigan" ;tag=7531 Call-ID: 1180028884-3040-TOPLAP@10.4.20.184 CSeq: 897 ACK Max-Forwards: 20 User-Agent: Uplink Skype2Sip 1.1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Itchy*CLI> <--- SIP read from 10.4.20.184:5060 ---> INVITE sip:1000@10.4.20.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK233040 To: From: "Joe S. Phigan" ;tag=7531 Call-ID: 1180028884-3040-TOPLAP@10.4.20.184 CSeq: 898 INVITE Max-Forwards: 20 User-Agent: Uplink Skype2Sip 1.1 Contact: Proxy-Authorization: Digest username="7101",realm="asterisk",nonce="0163fef3",uri="sip:1000@10.4.20.111",response="fe2758259bfc973c20f55630aa51dcf8",opaque="" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 284 v=0 o=- 836899966 836899988 IN IP4 10.4.20.184 s=Uplink Skype2Sip c=IN IP4 10.4.20.184 t=0 0 m=audio 10000 RTP/AVP 0 8 3 13 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:13 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (14 headers 14 lines) --- Sending to 10.4.20.184 : 5060 (NAT) Using INVITE request as basis request - 1180028884-3040-TOPLAP@10.4.20.184 Found user '7101' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 13 Found RTP audio format 101 Peer audio RTP is at port 10.4.20.184:10000 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format GSM for ID 3 Found description format CN for ID 13 Found description format telephone-event for ID 101 Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.4.20.184:10000 Looking for 1000 in fromskype (domain 10.4.20.111) list_route: hop: <--- Transmitting (NAT) to 10.4.20.184:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK233040;received=10.4.20.184;rport=5060 From: "Joe S. Phigan" ;tag=7531 To: Call-ID: 1180028884-3040-TOPLAP@10.4.20.184 CSeq: 898 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [1000@fromskype:1] Dial("SIP/7101-081d70b0", "Zap/2&SIP/101&SIP/102&SIP/103") in new stack [May 24 11:10:47] WARNING[13176]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) Really destroying SIP dialog '709d18be6b66c93a0874eb542a8d7f8e@127.0.1.1' Method: INVITE [May 24 11:10:47] WARNING[13176]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Really destroying SIP dialog '3052319774cbdbe91384b7494f849786@127.0.1.1' Method: INVITE [May 24 11:10:47] WARNING[13176]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Audio is at 10.4.20.111 port 10002 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 (NAT) to 10.4.20.190:5060: INVITE sip:103@10.4.20.190:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK04fde0c1;rport From: "From Skype" ;tag=as4bfc4222 To: Contact: Call-ID: 483533dc355c3c356e728cf63f9df4cb@10.4.20.111 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 24 May 2007 18:10:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 271 v=0 o=root 5910 5910 IN IP4 10.4.20.111 s=session c=IN IP4 10.4.20.111 t=0 0 m=audio 10002 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=sendrecv --- -- Called 103 Itchy*CLI> <--- SIP read from 10.4.20.190:5060 ---> SIP/2.0 100 Trying Call-ID: 483533dc355c3c356e728cf63f9df4cb@10.4.20.111 From: "From Skype";tag=as4bfc4222 To: ;tag=3c0f0-98612663 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK04fde0c1;rport Contact: Cordless Server: DTA SIP/0.12.35 DNLD/1.1.13 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Itchy*CLI> <--- SIP read from 10.4.20.190:5060 ---> SIP/2.0 180 Ringing Call-ID: 483533dc355c3c356e728cf63f9df4cb@10.4.20.111 From: "From Skype";tag=as4bfc4222 To: ;tag=3c0f0-98612663 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK04fde0c1;rport Contact: Cordless Server: DTA SIP/0.12.35 DNLD/1.1.13 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/103-081db1e0 is ringing <--- Transmitting (NAT) to 10.4.20.184:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK233040;received=10.4.20.184;rport=5060 From: "Joe S. Phigan" ;tag=7531 To: ;tag=as6a0aa85e Call-ID: 1180028884-3040-TOPLAP@10.4.20.184 CSeq: 898 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Itchy*CLI> <--- SIP read from 10.4.20.190:5060 ---> H <-------------> --- (1 headers 0 lines) --- Itchy*CLI> <--- SIP read from 10.4.20.190:5060 ---> SIP/2.0 200 OK Call-ID: 483533dc355c3c356e728cf63f9df4cb@10.4.20.111 From: "From Skype";tag=as4bfc4222 To: ;tag=3c0f0-98612663 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK04fde0c1;rport Contact: Cordless Supported: timer,100rel Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS,PRACK Server: DTA SIP/0.12.35 DNLD/1.1.13 Content-Type: application/sdp Content-Length: 132 v=0 o=103 0 0 IN IP4 10.4.20.190 s=- c=IN IP4 10.4.20.190 t=0 0 m=audio 8006 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 <-------------> --- (12 headers 7 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.4.20.190:8006 Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.4.20.190:8006 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.4.20.190, port 5060 Transmitting (NAT) to 10.4.20.190:5060: ACK sip:103@10.4.20.190:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK1744bff9;rport From: "From Skype" ;tag=as4bfc4222 To: ;tag=3c0f0-98612663 Contact: Call-ID: 483533dc355c3c356e728cf63f9df4cb@10.4.20.111 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/103-081db1e0 answered SIP/7101-081d70b0 Audio is at 10.4.20.111 port 10014 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 10.4.20.184:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK233040;received=10.4.20.184;rport=5060 From: "Joe S. Phigan" ;tag=7531 To: ;tag=as6a0aa85e Call-ID: 1180028884-3040-TOPLAP@10.4.20.184 CSeq: 898 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 271 v=0 o=root 5910 5910 IN IP4 10.4.20.111 s=session c=IN IP4 10.4.20.111 t=0 0 m=audio 10014 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv <------------> -- Packet2Packet bridging SIP/7101-081d70b0 and SIP/103-081db1e0 Itchy*CLI> <--- SIP read from 10.4.20.184:5060 ---> ACK sip:1000@10.4.20.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK243040 To: ;tag=as6a0aa85e From: "Joe S. Phigan" ;tag=7531 Call-ID: 1180028884-3040-TOPLAP@10.4.20.184 CSeq: 898 ACK Max-Forwards: 20 User-Agent: Uplink Skype2Sip 1.1 Proxy-Authorization: Digest username="7101",realm="asterisk",nonce="0163fef3",uri="sip:1000@10.4.20.111",response="fe2758259bfc973c20f55630aa51dcf8",opaque="" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '1180028882-3040-TOPLAP@10.4.20.184' Method: ACK Really destroying SIP dialog '3ad373b10b6946c13dea557f3bbd52b8@127.0.1.1' Method: REGISTER Itchy*CLI> <--- SIP read from 10.4.20.190:5060 ---> H <-------------> --- (1 headers 0 lines) --- Itchy*CLI> <--- SIP read from 10.4.20.184:5060 ---> BYE sip:1000@10.4.20.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK253040 To: ;tag=as6a0aa85e From: "Joe S. Phigan" ;tag=7531 Call-ID: 1180028884-3040-TOPLAP@10.4.20.184 CSeq: 899 BYE Max-Forwards: 20 User-Agent: Uplink Skype2Sip 1.1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 10.4.20.184 : 5060 (NAT) <--- Transmitting (NAT) to 10.4.20.184:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK253040;received=10.4.20.184;rport=5060 From: "Joe S. Phigan" ;tag=7531 To: ;tag=as6a0aa85e Call-ID: 1180028884-3040-TOPLAP@10.4.20.184 CSeq: 899 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '483533dc355c3c356e728cf63f9df4cb@10.4.20.111' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.4.20.190, port 5060 Reliably Transmitting (NAT) to 10.4.20.190:5060: BYE sip:103@10.4.20.190:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK557cc0fd;rport From: "From Skype" ;tag=as4bfc4222 To: ;tag=3c0f0-98612663 Contact: all-ID: 483533dc355c3c356e728cf63f9df4cb@10.4.20.111 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (fromskype, 1000, 1) exited non-zero on 'SIP/7101-081d70b0' Itchy*CLI> <--- SIP read from 10.4.20.190:5060 ---> SIP/2.0 200 OK Call-ID: 483533dc355c3c356e728cf63f9df4cb@10.4.20.111 From: "From Skype";tag=as4bfc4222 To: ;tag=3c0f0-98612663 CSeq: 103 BYE Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK557cc0fd;rport Contact: Cordless Server: DTA SIP/0.12.35 DNLD/1.1.13 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '483533dc355c3c356e728cf63f9df4cb@10.4.20.111' Method: INVITE Really destroying SIP dialog '1180028884-3040-TOPLAP@10.4.20.184' Method: BYE