Sip read: INVITE sip:0390177777@192.168.0.3:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.253.5:5060 From: "anonymous" ;tag=80DC105C-25F9 To: Date: Tue, 23 Mar 2004 05:47:06 GMT Call-ID: 5A25869F-7BC411D8-8818A715-58713D7B@192.168.253.5 Supported: 100rel Cisco-Guid: 1512328630-2076447192-2283185941-1483816315 User-Agent: Cisco-SIPGateway/IOS-12.x CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 1080020826 Contact: Expires: 180 Content-Type: application/sdp Content-Length: 213 v=0 o=CiscoSystemsSIP-GW-UserAgent 315 9157 IN IP4 192.168.253.5 s=SIP Call c=IN IP4 192.168.253.5 t=0 0 m=audio 19034 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 16 headers, 9 lines Using latest request as basis request Sending to 192.168.253.5 : 5060 (non-NAT) Found audio format UNKN Found audio format UNKN Found description format PCMU Found description format telephone-event Capabilities: us - 524302, them - 4/0, combined - 4 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 0390177777 in default list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.253.5:5060 From: "anonymous" ;tag=80DC105C-25F9 To: ;tag=as62a9d8ab Call-ID: 5A25869F-7BC411D8-8818A715-58713D7B@192.168.253.5 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.253.5:5060 -- Executing Goto("SIP/-08161700", "firstel|s|1") in new stack -- Goto (firstel,s,1) -- Executing Answer("SIP/-08161700", "") in new stack We're at 192.168.0.3 port 15684 Answering with capability 4 Answering with non-codec capability 1 Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.253.5:5060 From: "anonymous" ;tag=80DC105C-25F9 To: ;tag=as62a9d8ab Call-ID: 5A25869F-7BC411D8-8818A715-58713D7B@192.168.253.5 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 187 v=0 o=root 18457 18457 IN IP4 192.168.0.3 s=session c=IN IP4 192.168.0.3 t=0 0 m=audio 15684 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 to 192.168.253.5:5060 -- Executing SetMusicOnHold("SIP/-08161700", "default") in new stack -- Executing Playback("SIP/-08161700", "pls-wait-connect-call") in new stack -- Playing 'pls-wait-connect-call' (language 'en') asterisk*CLI> Sip read: ACK sip:0390177777@192.168.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.253.5:5060 From: "anonymous" ;tag=80DC105C-25F9 To: ;tag=as62a9d8ab Date: Tue, 23 Mar 2004 05:47:06 GMT Call-ID: 5A25869F-7BC411D8-8818A715-58713D7B@192.168.253.5 Max-Forwards: 6 Content-Length: 0 CSeq: 101 ACK 9 headers, 0 lines -- Executing Queue("SIP/-08161700", "Firstel-Direct") in new stack -- Called 3001 -- Executing Dial("Local/3001@default-61e5,2", "SIP/BYEXTENSION") in new stack We're at 192.168.0.3 port 17958 Answering with capability 2 Answering with capability 4 Answering with capability 8 Answering with non-codec capability 1 12 headers, 11 lines Reliably Transmitting: INVITE sip:3001@192.168.1.97 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK266604da From: "anonymous" ;tag=as495f04bf To: Contact: Call-ID: 16b80067118ce96869d95c5b539afa94@192.168.0.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 23 Mar 2004 05:43:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 18460 18460 IN IP4 192.168.0.3 s=session c=IN IP4 192.168.0.3 t=0 0 m=audio 17958 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 (no NAT) to 192.168.1.97:5060 -- Called 3001