IP read from 192.168.7.57:5060: INVITE sip:61312310974215696@192.168.7.3 SIP/2.0 CSeq: 1 INVITE Call-ID: call-80AC2B10-21BF-D311-0018-13@192.168.7.57 Contact: Content-Type: application/sdp From: ;tag=c0a80739-10 Session-GUID: 859321957-1631085668-825439845-962934528 To: Via: SIP/2.0/UDP 192.168.7.57;branch=z9hG4bK-tenor-c0a8-0739-0257 Content-Length: 224 User-Agent: Quintum/1.0.0 Quintum: 0c01030b0233300501000717000000000000000f006c0c8281808035353534313030331301000f0b413031322d313030313632 Max-Forwards: 70 v=0 o=Quintum 25 8 IN IP4 192.168.7.57 s=VoipCall c=IN IP4 192.168.7.57 t=0 0 m=audio 10260 RTP/AVP 18 8 101 c=IN IP4 192.168.7.57 a=rtpmap:18 g729/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:101 telephone-event/8000/1 --- (13 headers 10 lines)--- Using INVITE request as basis request - call-80AC2B10-21BF-D311-0018-13@192.168.7.57 Sending to 192.168.7.57 : 5060 (non-NAT) Found peer '55521003' Reliably Transmitting (no NAT) to 192.168.7.57:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.7.57;branch=z9hG4bK-tenor-c0a8-0739-0257;received=192.168.7.57 From: ;tag=c0a80739-10 To: ;tag=as53eb3e7d Call-ID: call-80AC2B10-21BF-D311-0018-13@192.168.7.57 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="49ac98d0" Content-Length: 0 --- Scheduling destruction of call 'call-80AC2B10-21BF-D311-0018-13@192.168.7.57' in 15000 ms haddock*CLI> <-- SIP read from 192.168.7.57:5060: ACK sip:61312310974215696@192.168.7.3 SIP/2.0 CSeq: 1 ACK Call-ID: call-80AC2B10-21BF-D311-0018-13@192.168.7.57 Contact: From: ;tag=c0a80739-10 Session-GUID: 859321957-1631085668-825439845-962934528 To: ;tag=as53eb3e7d Via: SIP/2.0/UDP 192.168.7.57;branch=z9hG4bK-tenor-c0a8-0739-0257 User-Agent: Quintum/1.0.0 Quintum: 0c01030b0233300501000717000000000000000f006c0c8281808035353534313030331301000f0b413031322d313030313632 Max-Forwards: 70 --- (11 headers 0 lines)--- haddock*CLI> <-- SIP read from 192.168.7.57:5060: INVITE sip:61312310974215696@192.168.7.3 SIP/2.0 CSeq: 2 INVITE Call-ID: call-80AC2B10-21BF-D311-0018-13@192.168.7.57 Contact: Content-Type: application/sdp From: ;tag=c0a80739-10 Proxy-Authorization: Digest realm="asterisk", nonce="49ac98d0", username="55521003", uri="sip:61312310974215696@192.168.7.3", response="4128f5e1649c44720479f8d00ef43061" Session-GUID: 859321957-1631085668-825439845-962934528 To: Via: SIP/2.0/UDP 192.168.7.57;branch=z9hG4bK-tenor-c0a8-0739-0258 Content-Length: 224 User-Agent: Quintum/1.0.0 Quintum: 0c01030b0233300501000717000000000000000f006c0c8281808035353534313030331301000f0b413031322d313030313632 Max-Forwards: 70 v=0 o=Quintum 25 8 IN IP4 192.168.7.57 s=VoipCall c=IN IP4 192.168.7.57 t=0 0 m=audio 10260 RTP/AVP 18 8 101 c=IN IP4 192.168.7.57 a=rtpmap:18 g729/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:101 telephone-event/8000/1 --- (14 headers 10 lines)--- Using INVITE request as basis request - call-80AC2B10-21BF-D311-0018-13@192.168.7.57 Sending to 192.168.7.57 : 5060 (non-NAT) Found peer '55521003' Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.7.57:10260 Found description format g729 Found description format pcma Found description format telephone-event Capabilities: us - 0x108 (alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 61312310974215696 in 5552 (domain 192.168.7.3) list_route: hop: Transmitting (no NAT) to 192.168.7.57:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.7.57;branch=z9hG4bK-tenor-c0a8-0739-0258;received=192.168.7.57 From: ;tag=c0a80739-10 To: Call-ID: call-80AC2B10-21BF-D311-0018-13@192.168.7.57 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- -- Executing Macro("SIP/55521003-d0a6", "pstn|61312310974215696") in new stack -- Executing Dial("SIP/55521003-d0a6", "SIP/61312310974215696@193.19.106.4") in new stack We're at 192.168.7.3 port 15706 Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 193.19.106.4:5060: INVITE sip:61312310974215696@193.19.106.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.3:5060;branch=z9hG4bK5fd34025;rport From: "55521003" ;tag=as0e6720d8 To: Contact: Call-ID: 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 02 Dec 2005 11:42:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 259 v=0 o=root 24334 24334 IN IP4 192.168.7.3 s=session c=IN IP4 192.168.7.3 t=0 0 m=audio 15706 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=noa=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 61312310974215696@193.19.106.4 haddock*CLI> <-- SIP read from 193.19.106.4:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.7.3:5060;branch=z9hG4bK5fd34025;rport From: "55521003" ;tag=as0e6720d8 To: ;tag=64E62FD0-24BF Date: Fri, 02 Dec 2005 11:42:52 GMT Call-ID: 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 --- (10 headers 0 lines)--- haddock*CLI> <-- SIP read from 193.19.106.4:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.7.3:5060;branch=z9hG4bK5fd34025;rport From: "55521003" ;tag=as0e6720d8 To: ;tag=64E62FD0-24BF Date: Fri, 02 Dec 2005 11:42:52 GMT Call-ID: 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: UPDATE Allow-Events: telephone-event Contact: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 258 v=0 o=CiscoSystemsSIP-GW-UserAgent 6908 2101 IN IP4 193.19.106.4 s=SIP Call c=IN IP4 193.19.106.4 t=0 0 m=audio 18120 RTP/AVP 18 101 c=IN IP4 193.19.106.4 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 --- (14 headers 11 lines)--- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 193.19.106.4:18120 Found description format G729 Found description format telephone-event Capabilities: us - 0x108 (alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- SIP/193.19.106.4-cb74 is making progress passing it to SIP/55521003-d0a6 We're at 192.168.7.3 port 12340 Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (no NAT) to 192.168.7.57:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.7.57;branch=z9hG4bK-tenor-c0a8-0739-0258;received=192.168.7.57 From: ;tag=c0a80739-10 To: ;tag=as2ad33411 Call-ID: call-80AC2B10-21BF-D311-0018-13@192.168.7.57 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 24334 24334 IN IP4 192.168.7.3 s=session c=IN IP4 192.168.7.3 t=0 0 m=audio 12340 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=noa=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- haddock*CLI> <-- SIP read from 193.19.106.4:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.3:5060;branch=z9hG4bK5fd34025;rport From: "55521003" ;tag=as0e6720d8 To: ;tag=64E62FD0-24BF Date: Fri, 02 Dec 2005 11:42:52 GMT Call-ID: 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Content-Type: application/sdp Content-Length: 258 v=0 o=CiscoSystemsSIP-GW-UserAgent 6908 2101 IN IP4 193.19.106.4 s=SIP Call c=IN IP4 193.19.106.4 t=0 0 m=audio 18120 RTP/AVP 18 101 c=IN IP4 193.19.106.4 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 --- (13 headers 11 lines)--- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 193.19.106.4:18120 Found description format G729 Found description format telephone-event Capabilities: us - 0x108 (alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 193.19.106.4, port 5060 Transmitting (no NAT) to 193.19.106.4:5060: ACK sip:61312310974215696@193.19.106.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.3:5060;branch=z9hG4bK07df3f88;rport From: "55521003" ;tag=as0e6720d8 To: ;tag=64E62FD0-24BF Contact: Call-ID: 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/193.19.106.4-cb74 answered SIP/55521003-d0a6 We're at 192.168.7.3 port 12340 Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.7.57:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.57;branch=z9hG4bK-tenor-c0a8-0739-0258;received=192.168.7.57 From: ;tag=c0a80739-10 To: ;tag=as2ad33411 Call-ID: call-80AC2B10-21BF-D311-0018-13@192.168.7.57 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 24334 24335 IN IP4 192.168.7.3 s=session c=IN IP4 192.168.7.3 t=0 0 m=audio 12340 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=noa=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/55521003-d0a6 and SIP/193.19.106.4-cb74 set_destination: Parsing for address/port to send to set_destination: set destination to 193.19.106.4, port 5060 We're at 192.168.7.3 port 15706 Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 193.19.106.4:5060: INVITE sip:61312310974215696@193.19.106.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.3:5060;branch=z9hG4bK500494ea;rport From: "55521003" ;tag=as0e6720d8 To: ;tag=64E62FD0-24BF Contact: Call-ID: 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 261 v=0 o=root 24334 24335 IN IP4 192.168.7.57 s=session c=IN IP4 192.168.7.57 t=0 0 m=audio 10260 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=noa=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- haddock*CLI> <-- SIP read from 193.19.106.4:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.3:5060;branch=z9hG4bK500494ea;rport From: "55521003" ;tag=as0e6720d8 To: ;tag=64E62FD0-24BF Date: Fri, 02 Dec 2005 11:42:59 GMT Call-ID: 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Content-Type: application/sdp Content-Length: 258 v=0 o=CiscoSystemsSIP-GW-UserAgent 6908 2102 IN IP4 193.19.106.4 s=SIP Call c=IN IP4 193.19.106.4 t=0 0 m=audio 18120 RTP/AVP 18 101 c=IN IP4 193.19.106.4 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 --- (13 headers 11 lines)--- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 193.19.106.4:18120 Found description format G729 Found description format telephone-event Capabilities: us - 0x108 (alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 193.19.106.4, port 5060 Transmitting (no NAT) to 193.19.106.4:5060: ACK sip:61312310974215696@193.19.106.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.3:5060;branch=z9hG4bK67a261bc;rport From: "55521003" ;tag=as0e6720d8 To: ;tag=64E62FD0-24BF Contact: Call-ID: 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- haddock*CLI> <-- SIP read from 192.168.7.57:5060: ACK sip:61312310974215696@192.168.7.3 SIP/2.0 CSeq: 2 ACK Call-ID: call-80AC2B10-21BF-D311-0018-13@192.168.7.57 Contact: From: ;tag=c0a80739-10 Proxy-Authorization: Digest realm="asterisk", nonce="49ac98d0", username="55521003", uri="sip:61312310974215696@192.168.7.3", response="4128f5e1649c44720479f8d00ef43061" Session-GUID: 859321957-1631085668-825439845-962934528 To: ;tag=as2ad33411 Via: SIP/2.0/UDP 192.168.7.57;branch=z9hG4bK-tenor-c0a8-0739-0258 User-Agent: Quintum/1.0.0 Quintum: 0c01030b0233300501000717000000000000000f006c0c8281808035353534313030331301000f0b413031322d313030313632 Max-Forwards: 70 --- (12 headers 0 lines)--- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.7.57, port 5060 We're at 192.168.7.3 port 12340 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 192.168.7.57:5060: INVITE sip:55541003@192.168.7.57 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.3:5060;branch=z9hG4bK4ed4e693;rport From: ;tag=as2ad33411 To: ;tag=c0a80739-10 Contact: Call-ID: call-80AC2B10-21BF-D311-0018-13@192.168.7.57 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 237 v=0 o=root 24334 24336 IN IP4 193.19.106.4 s=session c=IN IP4 193.19.106.4 t=0 0 m=audio 18120 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=noa=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- haddock*CLI> <-- SIP read from 192.168.7.57:5060: SIP/2.0 200 OK CSeq: 102 INVITE Call-ID: call-80AC2B10-21BF-D311-0018-13@192.168.7.57 Contact: Content-Type: application/sdp From: ;tag=as2ad33411 To: ;tag=c0a80739-10 Via: SIP/2.0/UDP 192.168.7.3:5060;branch=z9hG4bK4ed4e693;rport Content-Length: 202 User-Agent: Quintum/1.0.0 v=0 o=Quintum 26 24336 IN IP4 192.168.7.57 s=VoipCall c=IN IP4 192.168.7.57 t=0 0 m=audio 10260 RTP/AVP 18 101 c=IN IP4 192.168.7.57 a=rtpmap:18 g729/8000/1 a=rtpmap:101 telephone-event/8000/1 --- (10 headers 9 lines)--- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.7.57:10260 Found description format g729 Found description format telephone-event Capabilities: us - 0x108 (alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.7.57, port 5060 Transmitting (no NAT) to 192.168.7.57:5060: ACK sip:55541003@192.168.7.57 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.3:5060;branch=z9hG4bK08842020;rport From: ;tag=as2ad33411 To: ;tag=c0a80739-10 Contact: Call-ID: call-80AC2B10-21BF-D311-0018-13@192.168.7.57 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- haddock*CLI> <-- SIP read from 193.19.106.4:51162: INVITE sip:55521003@192.168.7.3:5060 SIP/2.0 Via: SIP/2.0/UDP 193.19.106.4:5060;x-route-tag="cid:asterisk_ip@192.168.7.3";branch=z9hG4bKF43 From: ;tag=64E62FD0-24BF To: "55521003" ;tag=as0e6720d8 Date: Fri, 02 Dec 2005 11:43:05 GMT Call-ID: 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 2604049937-1650397658-3137797698-3902329413 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: ;party=calling;screen=no;privacy=off Timestamp: 1133523785 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 183 v=0 o=CiscoSystemsSIP-GW-UserAgent 6908 2103 IN IP4 193.19.106.4 s=SIP Call c=IN IP4 193.19.106.4 t=0 0 m=audio 18120 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- (20 headers 8 lines)--- Using INVITE request as basis request - 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 Sending to 193.19.106.4 : 5060 (non-NAT) Found RTP audio format 8 Peer audio RTP is at port 193.19.106.4:18120 Found description format PCMA Capabilities: us - 0x108 (alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) We're at 192.168.7.3 port 15706 Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 193.19.106.4:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 193.19.106.4:5060;x-route-tag="cid:asterisk_ip@192.168.7.3";branch=z9hG4bKF43;received=193.19.106.4 From: ;tag=64E62FD0-24BF To: "55521003" ;tag=as0e6720d8 Call-ID: 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 205 v=0 o=root 24334 24336 IN IP4 192.168.7.57 s=session c=IN IP4 192.168.7.57 t=0 0 m=audio 10260 RTP/AVP 18 8 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=noa=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.7.57, port 5060 We're at 192.168.7.3 port 12340 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 192.168.7.57:5060: INVITE sip:55541003@192.168.7.57 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.3:5060;branch=z9hG4bK086c64d0;rport From: ;tag=as2ad33411 To: ;tag=c0a80739-10 Contact: Call-ID: call-80AC2B10-21BF-D311-0018-13@192.168.7.57 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 216 v=0 o=root 24334 24337 IN IP4 193.19.106.4 s=session c=IN IP4 193.19.106.4 t=0 0 m=audio 18120 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- haddock*CLI> <-- SIP read from 193.19.106.4:53617: ACK sip:55521003@192.168.7.3:5060 SIP/2.0 Via: SIP/2.0/UDP 193.19.106.4:5060;x-route-tag="cid:asterisk_ip@192.168.7.3";branch=z9hG4bK1518 From: ;tag=64E62FD0-24BF To: "55521003" ;tag=as0e6720d8 Date: Fri, 02 Dec 2005 11:43:05 GMT Call-ID: 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 Max-Forwards: 6 Content-Length: 0 CSeq: 101 ACK --- (9 headers 0 lines)--- haddock*CLI> <-- SIP read from 192.168.7.57:5060: SIP/2.0 200 OK CSeq: 103 INVITE Call-ID: call-80AC2B10-21BF-D311-0018-13@192.168.7.57 Contact: Content-Type: application/sdp From: ;tag=as2ad33411 To: ;tag=c0a80739-10 Via: SIP/2.0/UDP 192.168.7.3:5060;branch=z9hG4bK086c64d0;rport Content-Length: 200 User-Agent: Quintum/1.0.0 v=0 o=Quintum 27 24337 IN IP4 192.168.7.57 s=VoipCall c=IN IP4 192.168.7.57 t=0 0 m=audio 10260 RTP/AVP 8 101 c=IN IP4 192.168.7.57 a=rtpmap:8 pcma/8000/1 a=rtpmap:101 telephone-event/8000/1 --- (10 headers 9 lines)--- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.7.57:10260 Found description format pcma Found description format telephone-event Capabilities: us - 0x108 (alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.7.57, port 5060 Transmitting (no NAT) to 192.168.7.57:5060: ACK sip:55541003@192.168.7.57 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.3:5060;branch=z9hG4bK66df4955;rport From: ;tag=as2ad33411 To: ;tag=c0a80739-10 Contact: Call-ID: call-80AC2B10-21BF-D311-0018-13@192.168.7.57 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- haddock*CLI> <-- SIP read from 193.19.106.4:55502: INVITE sip:55521003@192.168.7.3:5060 SIP/2.0 Via: SIP/2.0/UDP 193.19.106.4:5060;x-route-tag="cid:asterisk_ip@192.168.7.3";branch=z9hG4bK216 From: ;tag=64E62FD0-24BF To: "55521003" ;tag=as0e6720d8 Date: Fri, 02 Dec 2005 11:43:25 GMT Call-ID: 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 2604049937-1650397658-3137797698-3902329413 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 102 INVITE Max-Forwards: 6 Remote-Party-ID: ;party=calling;screen=no;privacy=off Timestamp: 1133523805 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 183 v=0 o=CiscoSystemsSIP-GW-UserAgent 6908 2105 IN IP4 193.19.106.4 s=SIP Call c=IN IP4 193.19.106.4 t=0 0 m=audio 18120 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- (20 headers 8 lines)--- Using INVITE request as basis request - 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 Sending to 193.19.106.4 : 5060 (non-NAT) Found RTP audio format 8 Peer audio RTP is at port 193.19.106.4:18120 Found description format PCMA Capabilities: us - 0x108 (alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) We're at 192.168.7.3 port 15706 Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 193.19.106.4:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 193.19.106.4:5060;x-route-tag="cid:asterisk_ip@192.168.7.3";branch=z9hG4bK216;received=193.19.106.4 From: ;tag=64E62FD0-24BF To: "55521003" ;tag=as0e6720d8 Call-ID: 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 205 v=0 o=root 24334 24337 IN IP4 192.168.7.57 s=session c=IN IP4 192.168.7.57 t=0 0 m=audio 10260 RTP/AVP 18 8 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=noa=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- haddock*CLI> <-- SIP read from 193.19.106.4:55502: ACK sip:55521003@192.168.7.3:5060 SIP/2.0 Via: SIP/2.0/UDP 193.19.106.4:5060;x-route-tag="cid:asterisk_ip@192.168.7.3";branch=z9hG4bK1B7B From: ;tag=64E62FD0-24BF To: "55521003" ;tag=as0e6720d8 Date: Fri, 02 Dec 2005 11:43:25 GMT Call-ID: 2ac169b84722f5433a5da6327b5d0358@192.168.7.3 Max-Forwards: 6 Content-Length: 0 CSeq: 102 ACK --- (9 headers 0 lines)--- haddock*CLI> <-- SIP read from 192.168.7.57:5060: REGISTER sip:192.168.7.3 SIP/2.0 CSeq: 1148 REGISTER Call-ID: call-00B8782B-4CBE-D311-0018-0@192.168.7.57 Authorization: Digest realm="asterisk", nonce="2a86086d", username="55521003", uri="sip:192.168.7.3", response="6b835da80122282cee67fe0398cc2c93" Contact: Expires: 200 From: ;tag=c0a80739-f To: Via: SIP/2.0/UDP 192.168.7.57;branch=z9hG4bK-tenor-c0a8-0739-0256 User-Agent: Quintum/1.0.0 Max-Forwards: 70 --- (11 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.7.57 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.7.57:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.7.57;branch=z9hG4bK-tenor-c0a8-0739-0256;received=192.168.7.57 From: ;tag=c0a80739-f To: Call-ID: call-00B8782B-4CBE-D311-0018-0@192.168.7.57 CSeq: 1148 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.7.57:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.7.57;branch=z9hG4bK-tenor-c0a8-0739-0256;received=192.168.7.57 From: ;tag=c0a80739-f To: ;tag=as73a6222e Call-ID: call-00B8782B-4CBE-D311-0018-0@192.168.7.57 CSeq: 1148 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: WWW-Authenticate: Digest realm="asterisk", nonce="7d15ea94" Content-Length: 0 --- Scheduling destruction of call 'call-00B8782B-4CBE-D311-0018-0@192.168.7.57' in 15000 ms haddock*CLI> <-- SIP read from 192.168.7.57:5060: REGISTER sip:192.168.7.3 SIP/2.0 CSeq: 1149 REGISTER Call-ID: call-00B8782B-4CBE-D311-0018-0@192.168.7.57 Authorization: Digest realm="asterisk", nonce="7d15ea94", username="55521003", uri="sip:192.168.7.3", response="72aecf1b033c60730726cc46cc53316e" Contact: Expires: 200 From: ;tag=c0a80739-f To: Via: SIP/2.0/UDP 192.168.7.57;branch=z9hG4bK-tenor-c0a8-0739-0259 User-Agent: Quintum/1.0.0 Max-Forwards: 70 --- (11 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.7.57 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.7.57:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.7.57;branch=z9hG4bK-tenor-c0a8-0739-0259;received=192.168.7.57 From: ;tag=c0a80739-f To: Call-ID: call-00B8782B-4CBE-D311-0018-0@192.168.7.57 CSeq: 1149 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.7.57:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.57;branch=z9hG4bK-tenor-c0a8-0739-0259;received=192.168.7.57 From: ;tag=c0a80739-f To: ;tag=as73a6222e Call-ID: call-00B8782B-4CBE-D311-0018-0@192.168.7.57 CSeq: 1149 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Expires: 200 Contact: ;expires=200 Date: Fri, 02 Dec 2005 11:43:34 GMT Content-Length: 0 --- Scheduling destruction of call 'call-00B8782B-4CBE-D311-0018-0@192.168.7.57' in 15000 ms haddock*CLI>