Asterisk CVS-12/24/02-01:48:26, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-12 currently running on pbx (pid = 31226) pbx*CLI> Sip read: INVITE sip:4410001@192.168.1.10;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.7:5060 From: sip:ata1@192.168.1.10;tag=2867598075 To: Call-ID: 909601276@192.168.1.7 CSeq: 1 INVITE Contact: User-Agent: Cisco ATA v2.15 ata18x (020927a) Expires: 300 Content-Length: 255 Content-Type: application/sdp v=0 o=ata1 129291366 129291366 IN IP4 192.168.1.7 s=ATA186 Call c=IN IP4 192.168.1.7 t=0 0 m=audio 20000 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 11 lines Interface is eth0 IP Address is 192.168.1.10 Using latest request as basis request Sending to 192.168.1.7 : 5060 Capabilities: us - 14, them - 268, combined - 12 Transmitting: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.7:5060 From: sip:ata1@192.168.1.10;tag=2867598075 To: ;tag=43120866 Call-ID: 909601276@192.168.1.7 CSeq: 1 INVITE User-Agent: Asterisk PBX Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="10c1bdb2" Content-Length: 0 to 192.168.1.7:5060 pbx*CLI> Sip read: ACK sip:4410001@192.168.1.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.7:5060 From: sip:ata1@192.168.1.10;tag=2867598075 To: ;tag=43120866 Call-ID: 909601276@192.168.1.7 CSeq: 1 ACK User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 8 headers, 0 lines pbx*CLI> Sip read: INVITE sip:4410001@192.168.1.10;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.7:5060 From: sip:ata1@192.168.1.10;tag=2867598075 To: Call-ID: 909601276@192.168.1.7 CSeq: 2 INVITE Contact: User-Agent: Cisco ATA v2.15 ata18x (020927a) Proxy-Authorization: Digest username="ata1",realm="asterisk",nonce="10c1bdb2",uri="sip:4410001@192.168.1.10",response="3ca386c87a29073c533118d8364d975b" Expires: 300 Content-Length: 255 Content-Type: application/sdp v=0 o=ata1 129291367 129291367 IN IP4 192.168.1.7 s=ATA186 Call c=IN IP4 192.168.1.7 t=0 0 m=audio 20000 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 192.168.1.7 : 5060 Capabilities: us - 14, them - 268, combined - 12 Looking for 4410001 in home Transmitting: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.7:5060 From: sip:ata1@192.168.1.10;tag=2867598075 To: ;tag=2c5e5241 Call-ID: 909601276@192.168.1.7 CSeq: 2 INVITE User-Agent: Asterisk PBX Contact: Content-Length: 0 to 192.168.1.7:5060 pbx*CLI> == Accepting call on 'SIP/ata1-78af' (ata1) pbx*CLI> -- Executing Goto("SIP/ata1-78af", "fwd|BYEXTENSION|1") in new stack pbx*CLI> -- Goto (fwd,4410001,1) pbx*CLI> -- Executing StripMSD("SIP/ata1-78af", "2") in new stack pbx*CLI> -- Executing Dial("SIP/ata1-78af", "SIP/BYEXTENSION@fwdial") in new stack pbx*CLI> Interface is eth0 pbx*CLI> IP Address is 192.168.1.10 pbx*CLI> We're at 192.168.1.10 port 57034 pbx*CLI> Answering with capability 2 pbx*CLI> Answering with capability 4 pbx*CLI> Answering with capability 8 pbx*CLI> 10 headers, 11 lines pbx*CLI> XXX Need to handle Retransmitting XXX: INVITE sip:10001@192.246.69.247:5082 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=5761cc8c From: "ata1" ;tag=360abc66 Contact: To: Call-ID: 6385a77c0bf713056c74114f33a5fe4a@192.168.1.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Content-Type: application/sdp Content-Length: 236 v=0 o=root 31282 31282 IN IP4 192.168.1.10 s=session c=IN IP4 192.168.1.10 t=0 0 m=audio 57034 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 to 192.246.69.247:5082 pbx*CLI> -- Called 10001@fwdial pbx*CLI> Sip read: INVITE sip:10001@192.168.1.10:5060 SIP/2.0 v: SIP/2.0/UDP 192.246.69.247:34582;branch=PPCprotectedClient791605 f: "ata1" ;tag=360abc66 m: t: i: 6385a77c0bf713056c74114f33a5fe4a@192.168.1.10 CSeq: 102 INVITE c: application/sdp Record-Route: l: 216 v=0 o=X 31282 31282 IN IP4 192.246.69.247 s=session c=IN IP4 192.246.69.247 t=0 0 m=audio 10676 RTP/AVP 3 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 10 headers, 10 lines Transmitting: SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP 192.246.69.247:34582;branch=PPCprotectedClient791605 From: "ata1" ;tag=360abc66 To: ;tag=360abc66 Call-ID: 6385a77c0bf713056c74114f33a5fe4a@192.168.1.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Contact: Content-Length: 0 to 192.246.69.247:5082 pbx*CLI> Sip read: CANCEL sip:4410001@192.168.1.10;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.7:5060 From: sip:ata1@192.168.1.10;tag=2867598075 To: ;tag=2c5e5241 Call-ID: 909601276@192.168.1.7 CSeq: 2 CANCEL User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 8 headers, 0 lines Transmitting: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.7:5060 From: sip:ata1@192.168.1.10;tag=2867598075 To: ;tag=2c5e5241 Call-ID: 909601276@192.168.1.7 CSeq: 2 CANCEL User-Agent: Asterisk PBX Contact: Content-Length: 0 to 192.168.1.7:5060 pbx*CLI> XXX Need to handle Retransmitting XXX: CANCEL sip:10001@192.246.69.247:5082 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=5761cc8c From: "ata1" ;tag=360abc66 To: ;tag=360abc66 Call-ID: 6385a77c0bf713056c74114f33a5fe4a@192.168.1.10 CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0 to 192.246.69.247:5082 pbx*CLI> == Spawn extension (fwd, 10001, 2) exited non-zero on 'SIP/ata1-78af' pbx*CLI> Sip read: CANCEL sip:10001@192.168.1.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.246.69.247:34582;branch=PPCprotectedClient791605 From: "ata1" ;tag=360abc66 To: ;tag=360abc66 Call-ID: 6385a77c0bf713056c74114f33a5fe4a@192.168.1.10 CSeq: 102 CANCEL User-Agent: Asterisk PBX Record-Route: Content-Length: 0 9 headers, 0 lines Transmitting: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.246.69.247:34582;branch=PPCprotectedClient791605 From: "ata1" ;tag=360abc66 To: ;tag=360abc66 Call-ID: 6385a77c0bf713056c74114f33a5fe4a@192.168.1.10 CSeq: 102 CANCEL User-Agent: Asterisk PBX Contact: Content-Length: 0 to 192.246.69.247:5082 pbx*CLI> quit