<--- SIP read from 213.166.5.130:5060 ---> INVITE sip:01273808802@85.13.195.164 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKeb4.82a3dde4.0 Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK9FD8ED47F Remote-Party-ID: ;party=calling;screen=yes;privacy=off From: "07743898503" ;tag=28BEA090-1D6C To: Date: Fri, 14 Dec 2007 00:04:40 gmt Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 14 Timestamp: 1197590680 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 418 v=0 o=CiscoSystemsSIP-GW-UserAgent 4487 6672 IN IP4 213.166.5.134 s=SIP Call c=IN IP4 213.166.5.134 t=0 0 m=audio 22614 RTP/AVP 8 18 4 3 98 0 101 c=IN IP4 213.166.5.134 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=6.3;annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (22 headers 17 lines) --- == Using TOS bits 0 == Using CoS mark 5 Sending to 213.166.5.130 : 5060 (no NAT) Using INVITE request as basis request - FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 No user '07743898503' in SIP users list Found peer 'inbound' for '07743898503' from 213.166.5.130:5060 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.134:22614 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format GSM for ID 3 Found audio description format G726-32 for ID 98 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.134:22614 Looking for 01273808802 in main (domain 85.13.195.164) list_route: hop: <--- Transmitting (no NAT) to 213.166.5.130:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKeb4.82a3dde4.0;received=213.166.5.130 Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK9FD8ED47F Record-Route: From: "07743898503" ;tag=28BEA090-1D6C To: Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 CSeq: 101 INVITE User-Agent: Asterisk PBX SVN-trunk-r92855 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [01273808802@main:1] Answer("SIP/213.166.5.134-b6c2a160", "") in new stack Audio is at 85.13.195.164 port 18366 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 213.166.5.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKeb4.82a3dde4.0;received=213.166.5.130 Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK9FD8ED47F Record-Route: From: "07743898503" ;tag=28BEA090-1D6C To: ;tag=as007348af Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 CSeq: 101 INVITE User-Agent: Asterisk PBX SVN-trunk-r92855 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 272 v=0 o=root 983608028 983608028 IN IP4 85.13.195.164 s=Asterisk PBX SVN-trunk-r92855 c=IN IP4 85.13.195.164 t=0 0 m=audio 18366 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Executing [01273808802@main:2] Dial("SIP/213.166.5.134-b6c2a160", "SIP/01392213713@outbound") in new stack == Using TOS bits 0 == Using CoS mark 5 Audio is at 85.13.195.164 port 18716 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.166.5.135:5060: INVITE sip:01392213713@gk.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK62d9ec39;rport Max-Forwards: 70 From: "07743898503" ;tag=as6c85b7af To: Contact: Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r92855 Date: Fri, 14 Dec 2007 00:04:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 274 v=0 o=root 1017897569 1017897569 IN IP4 85.13.195.164 s=Asterisk PBX SVN-trunk-r92855 c=IN IP4 85.13.195.164 t=0 0 m=audio 18716 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 01392213713@outbound turing*CLI> <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK62d9ec39;rport=5060 From: "07743898503" ;tag=as6c85b7af To: ;tag=a3e149d53f0faf6bf88ecc254282dfd9.45bb Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="gk.magrathea.net", nonce="4761c9d7aef4305d8c15e55557646250a9e49c1e" Server: Sip EXpress router (0.8.99-dev (i386/linux)) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Transmitting (no NAT) to 213.166.5.135:5060: ACK sip:01392213713@gk.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK62d9ec39;rport Max-Forwards: 70 From: "07743898503" ;tag=as6c85b7af To: ;tag=a3e149d53f0faf6bf88ecc254282dfd9.45bb Contact: Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r92855 Content-Length: 0 --- Audio is at 85.13.195.164 port 18716 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.166.5.135:5060: INVITE sip:01392213713@gk.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK64b2294b;rport Max-Forwards: 70 From: "07743898503" ;tag=as6c85b7af To: Contact: Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-trunk-r92855 Proxy-Authorization: Digest username="ordersip", realm="gk.magrathea.net", algorithm=MD5, uri="sip:01392213713@gk.magrathea.net", nonce="4761c9d7aef4305d8c15e55557646250a9e49c1e", response="441b7ae56d981c76963c5c4e93353e48", opaque="" Date: Fri, 14 Dec 2007 00:04:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 274 v=0 o=root 1017897569 1017897570 IN IP4 85.13.195.164 s=Asterisk PBX SVN-trunk-r92855 c=IN IP4 85.13.195.164 t=0 0 m=audio 18716 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- turing*CLI> <--- SIP read from 213.166.5.130:5060 ---> ACK sip:01273808802@85.13.195.164:5060 SIP/2.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKeb4.82a3dde4.2 Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK9FD8EE51A From: ;tag=28BEA090-1D6C To: ;tag=as007348af Date: Fri, 14 Dec 2007 00:04:40 gmt Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 Max-Forwards: 14 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> --- (11 headers 0 lines) --- turing*CLI> <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK64b2294b;rport=5060 From: "07743898503" ;tag=as6c85b7af To: Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 CSeq: 103 INVITE Server: Sip EXpress router (0.8.99-dev (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- turing*CLI> <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK64b2294b;rport=5060 From: "07743898503" ;tag=as6c85b7af To: ;tag=39AE1F04-F6D Date: Fri, 14 Dec 2007 00:04:40 gmt Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 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: Record-Route: , Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 6149 2047 IN IP4 87.238.72.155 s=SIP Call c=IN IP4 87.238.72.155 t=0 0 m=audio 18144 RTP/AVP 8 101 c=IN IP4 87.238.72.155 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (15 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.155:18144 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.155:18144 -- SIP/outbound-08206e60 is making progress passing it to SIP/213.166.5.134-b6c2a160 set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.130, port 5060 Audio is at 85.13.195.164 port 18366 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.166.5.130:5060: INVITE sip:07743898503@213.166.5.134:5060 SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport Route: Max-Forwards: 70 From: ;tag=as007348af To: "07743898503" ;tag=28BEA090-1D6C Contact: Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r92855 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 272 v=0 o=root 983608028 983608029 IN IP4 87.238.72.155 s=Asterisk PBX SVN-trunk-r92855 c=IN IP4 87.238.72.155 t=0 0 m=audio 18144 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- turing*CLI> <--- SIP read from 213.166.5.130:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060 From: ;tag=as007348af To: "07743898503" ;tag=28BEA090-1D6C Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 CSeq: 102 INVITE Server: OpenSER (1.2.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- turing*CLI> <--- SIP read from 213.166.5.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060 From: ;tag=as007348af To: ;tag=28BEA090-1D6C Date: Fri, 14 Dec 2007 00:04:40 gmt Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=yes;privacy=off Contact: Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 4487 6673 IN IP4 213.166.5.134 s=SIP Call c=IN IP4 213.166.5.134 t=0 0 m=audio 22614 RTP/AVP 8 101 c=IN IP4 213.166.5.134 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.134:22614 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.134:22614 --- set_address_from_contact host '213.166.5.134' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.134, port 5060 Transmitting (no NAT) to 213.166.5.134:5060: ACK sip:07743898503@213.166.5.134:5060 SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK5b07a085;rport Max-Forwards: 70 From: ;tag=as007348af To: "07743898503" ;tag=28BEA090-1D6C Contact: Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r92855 Content-Length: 0 --- turing*CLI> <--- SIP read from 213.166.5.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060 From: ;tag=as007348af To: ;tag=28BEA090-1D6C Date: Fri, 14 Dec 2007 00:04:40 gmt Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=yes;privacy=off Contact: Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 4487 6673 IN IP4 213.166.5.134 s=SIP Call c=IN IP4 213.166.5.134 t=0 0 m=audio 22614 RTP/AVP 8 101 c=IN IP4 213.166.5.134 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.134:22614 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.134:22614 --- set_address_from_contact host '213.166.5.134' set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.134, port 5060 Transmitting (no NAT) to 213.166.5.134:5060: ACK sip:07743898503@213.166.5.134:5060 SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6c3fb7d2;rport Max-Forwards: 70 From: ;tag=as007348af To: "07743898503" ;tag=28BEA090-1D6C Contact: Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r92855 Content-Length: 0 --- turing*CLI> <--- SIP read from 213.166.5.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060 From: ;tag=as007348af To: ;tag=28BEA090-1D6C Date: Fri, 14 Dec 2007 00:04:40 gmt Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=yes;privacy=off Contact: Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 4487 6673 IN IP4 213.166.5.134 s=SIP Call c=IN IP4 213.166.5.134 t=0 0 m=audio 22614 RTP/AVP 8 101 c=IN IP4 213.166.5.134 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.134:22614 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.134:22614 --- set_address_from_contact host '213.166.5.134' set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.134, port 5060 Transmitting (no NAT) to 213.166.5.134:5060: ACK sip:07743898503@213.166.5.134:5060 SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK18572230;rport Max-Forwards: 70 From: ;tag=as007348af To: "07743898503" ;tag=28BEA090-1D6C Contact: Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r92855 Content-Length: 0 --- turing*CLI> <--- SIP read from 213.166.5.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060 From: ;tag=as007348af To: ;tag=28BEA090-1D6C Date: Fri, 14 Dec 2007 00:04:40 gmt Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=yes;privacy=off Contact: Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 4487 6673 IN IP4 213.166.5.134 s=SIP Call c=IN IP4 213.166.5.134 t=0 0 m=audio 22614 RTP/AVP 8 101 c=IN IP4 213.166.5.134 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.134:22614 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.134:22614 --- set_address_from_contact host '213.166.5.134' set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.134, port 5060 Transmitting (no NAT) to 213.166.5.134:5060: ACK sip:07743898503@213.166.5.134:5060 SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK67f02392;rport Max-Forwards: 70 From: ;tag=as007348af To: "07743898503" ;tag=28BEA090-1D6C Contact: Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r92855 Content-Length: 0 --- turing*CLI> <--- SIP read from 213.166.5.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060 From: ;tag=as007348af To: ;tag=28BEA090-1D6C Date: Fri, 14 Dec 2007 00:04:40 gmt Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=yes;privacy=off Contact: Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 4487 6673 IN IP4 213.166.5.134 s=SIP Call c=IN IP4 213.166.5.134 t=0 0 m=audio 22614 RTP/AVP 8 101 c=IN IP4 213.166.5.134 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.134:22614 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.134:22614 --- set_address_from_contact host '213.166.5.134' set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.134, port 5060 Transmitting (no NAT) to 213.166.5.134:5060: ACK sip:07743898503@213.166.5.134:5060 SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK3c5ae694;rport Max-Forwards: 70 From: ;tag=as007348af To: "07743898503" ;tag=28BEA090-1D6C Contact: Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r92855 Content-Length: 0 --- turing*CLI> <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK64b2294b;rport=5060 From: "07743898503" ;tag=as6c85b7af To: ;tag=39AE1F04-F6D Date: Fri, 14 Dec 2007 00:04:40 gmt Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 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: Record-Route: , Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 6149 2047 IN IP4 87.238.72.155 s=SIP Call c=IN IP4 87.238.72.155 t=0 0 m=audio 18144 RTP/AVP 8 101 c=IN IP4 87.238.72.155 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.155:18144 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.155:18144 --- set_address_from_contact host '87.238.72.155' list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.135, port 5060 Transmitting (no NAT) to 213.166.5.135:5060: ACK sip:*200441392213713@87.238.72.155:5060 SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK686cf6a3;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as6c85b7af To: ;tag=39AE1F04-F6D Contact: Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-trunk-r92855 Content-Length: 0 --- -- SIP/outbound-08206e60 answered SIP/213.166.5.134-b6c2a160 -- Native bridging SIP/213.166.5.134-b6c2a160 and SIP/outbound-08206e60 set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.135, port 5060 Audio is at 85.13.195.164 port 18716 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.166.5.135:5060: INVITE sip:*200441392213713@87.238.72.155:5060 SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK5faf1434;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as6c85b7af To: ;tag=39AE1F04-F6D Contact: Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 CSeq: 104 INVITE User-Agent: Asterisk PBX SVN-trunk-r92855 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 274 v=0 o=root 1017897569 1017897571 IN IP4 213.166.5.134 s=Asterisk PBX SVN-trunk-r92855 c=IN IP4 213.166.5.134 t=0 0 m=audio 22614 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- turing*CLI> <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK5faf1434;rport=5060 From: "07743898503" ;tag=as6c85b7af To: ;tag=39AE1F04-F6D Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 CSeq: 104 INVITE Server: Sip EXpress router (0.8.99-dev (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- turing*CLI> <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK5faf1434;rport=5060 From: "07743898503" ;tag=as6c85b7af To: ;tag=39AE1F04-F6D Date: Fri, 14 Dec 2007 00:04:51 gmt Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 Server: Cisco-SIPGateway/IOS-12.x CSeq: 104 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: , Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 6149 2048 IN IP4 87.238.72.155 s=SIP Call c=IN IP4 87.238.72.155 t=0 0 m=audio 18144 RTP/AVP 8 101 c=IN IP4 87.238.72.155 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.155:18144 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.155:18144 --- set_address_from_contact host '87.238.72.155' set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.135, port 5060 Transmitting (no NAT) to 213.166.5.135:5060: ACK sip:*200441392213713@87.238.72.155:5060 SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK33852ac0;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as6c85b7af To: ;tag=39AE1F04-F6D Contact: Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 CSeq: 104 ACK User-Agent: Asterisk PBX SVN-trunk-r92855 Content-Length: 0 --- turing*CLI> <--- SIP read from 213.166.5.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060 From: ;tag=as007348af To: ;tag=28BEA090-1D6C Date: Fri, 14 Dec 2007 00:04:40 gmt Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=yes;privacy=off Contact: Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 4487 6673 IN IP4 213.166.5.134 s=SIP Call c=IN IP4 213.166.5.134 t=0 0 m=audio 22614 RTP/AVP 8 101 c=IN IP4 213.166.5.134 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.134:22614 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.134:22614 --- set_address_from_contact host '213.166.5.134' set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.134, port 5060 Transmitting (no NAT) to 213.166.5.134:5060: ACK sip:07743898503@213.166.5.134:5060 SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK155ac28b;rport Max-Forwards: 70 From: ;tag=as007348af To: "07743898503" ;tag=28BEA090-1D6C Contact: Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r92855 Content-Length: 0 --- turing*CLI> <--- SIP read from 213.166.5.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060 From: ;tag=as007348af To: ;tag=28BEA090-1D6C Date: Fri, 14 Dec 2007 00:04:40 gmt Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=yes;privacy=off Contact: Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 4487 6673 IN IP4 213.166.5.134 s=SIP Call c=IN IP4 213.166.5.134 t=0 0 m=audio 22614 RTP/AVP 8 101 c=IN IP4 213.166.5.134 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 213.166.5.134:22614 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.5.134:22614 --- set_address_from_contact host '213.166.5.134' set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.134, port 5060 Transmitting (no NAT) to 213.166.5.134:5060: ACK sip:07743898503@213.166.5.134:5060 SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK241d5a13;rport Max-Forwards: 70 From: ;tag=as007348af To: "07743898503" ;tag=28BEA090-1D6C Contact: Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r92855 Content-Length: 0 --- turing*CLI> <--- SIP read from 213.166.5.130:5060 ---> BYE sip:01273808802@85.13.195.164:5060 SIP/2.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKbb4.9ba89a77.0 Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK9FD8F82687 From: ;tag=28BEA090-1D6C To: ;tag=as007348af Date: Fri, 14 Dec 2007 00:04:40 gmt Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 14 Timestamp: 1197590700 CSeq: 102 BYE Reason: Q.850;cause=127 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 213.166.5.130 : 5060 (no NAT) <--- Transmitting (no NAT) to 213.166.5.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKbb4.9ba89a77.0;received=213.166.5.130 Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK9FD8F82687 From: ;tag=28BEA090-1D6C To: ;tag=as007348af Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r92855 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.135, port 5060 Audio is at 85.13.195.164 port 18716 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.166.5.135:5060: INVITE sip:*200441392213713@87.238.72.155:5060 SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK790e4a2c;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as6c85b7af To: ;tag=39AE1F04-F6D Contact: Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 CSeq: 105 INVITE User-Agent: Asterisk PBX SVN-trunk-r92855 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 274 v=0 o=root 1017897569 1017897572 IN IP4 85.13.195.164 s=Asterisk PBX SVN-trunk-r92855 c=IN IP4 85.13.195.164 t=0 0 m=audio 18716 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Scheduling destruction of SIP dialog '25e865a71ceadeb207ee702953649f68@85.13.195.164' in 32000 ms (Method: INVITE) == Spawn extension (main, 01273808802, 2) exited non-zero on 'SIP/213.166.5.134-b6c2a160' turing*CLI> <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK790e4a2c;rport=5060 From: "07743898503" ;tag=as6c85b7af To: ;tag=39AE1F04-F6D Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 CSeq: 105 INVITE Server: Sip EXpress router (0.8.99-dev (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog 'FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134' Method: BYE turing*CLI> <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK790e4a2c;rport=5060 From: "07743898503" ;tag=as6c85b7af To: ;tag=39AE1F04-F6D Date: Fri, 14 Dec 2007 00:05:00 gmt Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 Server: Cisco-SIPGateway/IOS-12.x CSeq: 105 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: , Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 6149 2049 IN IP4 87.238.72.155 s=SIP Call c=IN IP4 87.238.72.155 t=0 0 m=audio 18144 RTP/AVP 8 101 c=IN IP4 87.238.72.155 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.155:18144 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.155:18144 --- set_address_from_contact host '87.238.72.155' set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.135, port 5060 Transmitting (no NAT) to 213.166.5.135:5060: ACK sip:*200441392213713@87.238.72.155:5060 SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK20d2886a;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as6c85b7af To: ;tag=39AE1F04-F6D Contact: Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 CSeq: 105 ACK User-Agent: Asterisk PBX SVN-trunk-r92855 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.135, port 5060 Reliably Transmitting (no NAT) to 213.166.5.135:5060: BYE sip:*200441392213713@87.238.72.155:5060 SIP/2.0 Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK042f8076;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as6c85b7af To: ;tag=39AE1F04-F6D Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 CSeq: 106 BYE User-Agent: Asterisk PBX SVN-trunk-r92855 Proxy-Authorization: Digest username="ordersip", realm="gk.magrathea.net", algorithm=MD5, uri="sip:*200441392213713@87.238.72.155:5060", nonce="4761c9d7aef4305d8c15e55557646250a9e49c1e", response="646daec82d933d60dd61366dd0a1d973", opaque="" Content-Length: 0 --- Scheduling destruction of SIP dialog '25e865a71ceadeb207ee702953649f68@85.13.195.164' in 32000 ms (Method: INVITE) turing*CLI> <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK042f8076;rport=5060 From: "07743898503" ;tag=as6c85b7af To: ;tag=39AE1F04-F6D Date: Fri, 14 Dec 2007 00:05:00 gmt Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 106 BYE <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '25e865a71ceadeb207ee702953649f68@85.13.195.164' Method: INVITE turing*CLI>