voicegate*CLI> <-- SIP read from 209.17.160.130:5060: INVITE sip:9193833430@24.211.242.232:5060 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.0 Via: SIP/2.0/UDP 216.143.130.87:5060;branch=z9hG4bK608318ec;rport=5060 From: "restricted" ;tag=as5b8f34d3 To: Contact: Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 17 Oct 2007 14:02:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 267 v=0 o=root 15912 15912 IN IP4 216.143.130.87 s=session c=IN IP4 216.143.130.87 t=0 0 m=audio 33974 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (14 headers 12 lines) --- Using INVITE request as basis request - 24459c045dfffa6c4ee6481b060dca33@216.143.130.87 Sending to 209.17.160.130 : 5060 (NAT) Found no matching peer or user for '209.17.160.130:5060' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 216.143.130.87:33974 Found description format PCMU Found description format G729 Found description format telephone-event Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 9193833430 in default (domain 24.211.242.232) Reliably Transmitting (NAT) to 209.17.160.130:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.0;received=209.17.160.130 Via: SIP/2.0/UDP 216.143.130.87:5060;branch=z9hG4bK608318ec;rport=5060 From: "restricted" ;tag=as5b8f34d3 To: ;tag=as7d2ff4e7 Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87 CSeq: 102 INVITE User-Agent: CallWeaver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Content-Length: 0 --- <-- SIP read from 209.17.160.130:5060: INVITE sip:s@24.211.242.232:5060 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.1 Via: SIP/2.0/UDP 216.143.130.87:5060;branch=z9hG4bK608318ec;rport=5060 From: "restricted" ;tag=as5b8f34d3 To: Contact: Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 17 Oct 2007 14:02:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 267 v=0 o=root 15912 15912 IN IP4 216.143.130.87 s=session c=IN IP4 216.143.130.87 t=0 0 m=audio 33974 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (14 headers 12 lines) --- Ignoring this INVITE request voicegate*CLI> <-- SIP read from 209.17.160.130:5060: ACK sip:9193833430@24.211.242.232:5060 SIP/2.0 Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.0 From: "restricted" ;tag=as5b8f34d3 Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87 To: ;tag=as7d2ff4e7 CSeq: 102 ACK User-Agent: Sip EXpress router(0.8.14 (i386/freebsd)) Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '24459c045dfffa6c4ee6481b060dca33@216.143.130.87' voicegate*CLI> <-- SIP read from 209.17.160.130:5060: INVITE sip:s@24.211.242.232:5060 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.1 Via: SIP/2.0/UDP 216.143.130.87:5060;branch=z9hG4bK608318ec;rport=5060 From: "restricted" ;tag=as5b8f34d3 To: Contact: Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 17 Oct 2007 14:02:14 GMT llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 267 v=0 o=root 15912 15912 IN IP4 216.143.130.87 s=session c=IN IP4 216.143.130.87 t=0 0 m=audio 33974 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (14 headers 12 lines) --- Using INVITE request as basis request - 24459c045dfffa6c4ee6481b060dca33@216.143.130.87 Sending to 209.17.160.130 : 5060 (NAT) Found no matching peer or user for '209.17.160.130:5060' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 216.143.130.87:33974 Found description format PCMU Found description format G729 Found description format telephone-event Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for s in default (domain 24.211.242.232) list_route: hop: Transmitting (NAT) to 209.17.160.130:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.1;received=209.17.160.130 Via: SIP/2.0/UDP 216.143.130.87:5060;branch=z9hG4bK608318ec;rport=5060 From: "restricted" ;tag=as5b8f34d3 To: Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87 CSeq: 102 INVITE User-Agent: CallWeaver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- Oct 17 06:01:12 WARNING[57835533]: channel.c:2283 set_format: Unable to find a codec translation path from g729 to slin Oct 17 06:01:12 WARNING[57835533]: channel.c:3698 tonepair_alloc: Unable to set 'SIP/216.143.130.87-40f14598' to signed linear format (write) Oct 17 06:01:12 ERROR[57835533]: generator.c:93 opbx_generator_activate: Generator activation failed Scheduling destruction of call '24459c045dfffa6c4ee6481b060dca33@216.143.130.87' in 32000 ms Reliably Transmitting (NAT) to 209.17.160.130:5060: SIP/2.0 603 Declined Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.1;received=209.17.160.130 Via: SIP/2.0/UDP 216.143.130.87:5060;branch=z9hG4bK608318ec;rport=5060 From: "restricted" ;tag=as5b8f34d3 To: ;tag=as7ddf6ec7 Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87 CSeq: 102 INVITE User-Agent: CallWeaver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- voicegate*CLI> <-- SIP read from 209.17.160.130:5060: ACK sip:s@24.211.242.232:5060 SIP/2.0 Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.1 From: "restricted" ;tag=as5b8f34d3 Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87 To: ;tag=as7ddf6ec7 CSeq: 102 ACK User-Agent: Sip EXpress router(0.8.14 (i386/freebsd)) Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '24459c045dfffa6c4ee6481b060dca33@216.143.130.87' voicegate*CLI>