Asterisk 1.4.13 call to 0845 working reinvite (87 group gateway). *CLI> <--- SIP read from 87.238.72.153:5060 ---> INVITE sip:08450045413@213.232.93.17 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK243c.88e6caf6.0 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bKB209AE1B71 From: "07743898503" ;tag=39BEDB98-E11 To: Date: Thu, 19 Feb 2009 16:33:33 gmt Call-ID: E2451AB2-FDD911DD-A531F3EC-65114225@87.238.72.149 Supported: 100rel,timer,replaces Min-SE: 1800 User-Agent: MSSGW(B) Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 14 Remote-Party-ID: ;party=calling;screen=yes;privacy=off Timestamp: 1235061213 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 403 v=0 o=CiscoSystemsSIP-GW-UserAgent 960 4612 IN IP4 87.238.72.149 s=SIP Call c=IN IP4 87.238.72.149 t=0 0 m=audio 17794 RTP/AVP 8 18 4 3 2 0 101 c=IN IP4 87.238.72.149 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (21 headers 17 lines) --- Sending to 87.238.72.153 : 5060 (no NAT) Using INVITE request as basis request - E2451AB2-FDD911DD-A531F3EC-65114225@87.238.72.149 Found no matching peer or user for '87.238.72.153:5060' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.149:17794 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format G723 for ID 4 Found description format GSM for ID 3 Found description format G726-32 for ID 2 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.149:17794 Looking for 08450045413 in main (domain 213.232.93.17) list_route: hop: <--- Transmitting (no NAT) to 87.238.72.153:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK243c.88e6caf6.0;received=87.238.72.153 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bKB209AE1B71 From: "07743898503" ;tag=39BEDB98-E11 To: Call-ID: E2451AB2-FDD911DD-A531F3EC-65114225@87.238.72.149 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [08450045413@main:1] Goto("SIP/87.238.72.149-081c29f0", "448450045413|1") in new stack -- Goto (main,448450045413,1) -- Executing [448450045413@main:1] Ringing("SIP/87.238.72.149-081c29f0", "") in new stack <--- Transmitting (no NAT) to 87.238.72.153:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK243c.88e6caf6.0;received=87.238.72.153 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bKB209AE1B71 From: "07743898503" ;tag=39BEDB98-E11 To: ;tag=as4e5059bd Call-ID: E2451AB2-FDD911DD-A531F3EC-65114225@87.238.72.149 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [448450045413@main:2] Answer("SIP/87.238.72.149-081c29f0", "") in new stack Audio is at 213.232.93.17 port 17504 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 87.238.72.153:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK243c.88e6caf6.0;received=87.238.72.153 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bKB209AE1B71 Record-Route: From: "07743898503" ;tag=39BEDB98-E11 To: ;tag=as4e5059bd Call-ID: E2451AB2-FDD911DD-A531F3EC-65114225@87.238.72.149 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 289 v=0 o=root 17226 17226 IN IP4 213.232.93.17 s=session c=IN IP4 213.232.93.17 t=0 0 m=audio 17504 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 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Executing [448450045413@main:3] Dial("SIP/87.238.72.149-081c29f0", "Local/dialOffice@main||ro") in new stack -- Called dialOffice@main -- Executing [dialOffice@main:1] Dial("Local/dialOffice@main-1fb8,2", "SIP/01392213713@outbound") in new stack Audio is at 213.232.93.17 port 18068 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 213.232.93.17:5060;branch=z9hG4bK6ef91d1d;rport From: "07743898503" ;tag=as4a4d4d0e To: Contact: Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 19 Feb 2009 16:33:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 242 v=0 o=root 17226 17226 IN IP4 213.232.93.17 s=session c=IN IP4 213.232.93.17 t=0 0 m=audio 18068 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 <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK6ef91d1d;rport=5060 From: "07743898503" ;tag=as4a4d4d0e To: ;tag=a3e149d53f0faf6bf88ecc254282dfd9.f497 Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="gk.magrathea.net", nonce="499d8b0ada57146340a9aa6d1ac6382fdd197079" 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 213.232.93.17:5060;branch=z9hG4bK6ef91d1d;rport From: "07743898503" ;tag=as4a4d4d0e To: ;tag=a3e149d53f0faf6bf88ecc254282dfd9.f497 Contact: Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Audio is at 213.232.93.17 port 18068 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 213.232.93.17:5060;branch=z9hG4bK54b66f44;rport From: "07743898503" ;tag=as4a4d4d0e To: Contact: Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="ordersip", realm="gk.magrathea.net", algorithm=MD5, uri="sip:01392213713@gk.magrathea.net", nonce="499d8b0ada57146340a9aa6d1ac6382fdd197079", response="2bad36dded7f68b7918a0b9b5f10550a", opaque="" Date: Thu, 19 Feb 2009 16:33:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 242 v=0 o=root 17226 17227 IN IP4 213.232.93.17 s=session c=IN IP4 213.232.93.17 t=0 0 m=audio 18068 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 --- <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK54b66f44;rport=5060 From: "07743898503" ;tag=as4a4d4d0e To: Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 103 INVITE Server: Sip EXpress router (0.8.99-dev (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 87.238.72.153:5060 ---> ACK sip:08450045413@213.232.93.17:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bK243c.88e6caf6.2 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bKB209AF22FD From: ;tag=39BEDB98-E11 To: ;tag=as4e5059bd Date: Thu, 19 Feb 2009 16:33:33 gmt Call-ID: E2451AB2-FDD911DD-A531F3EC-65114225@87.238.72.149 Max-Forwards: 14 CSeq: 101 ACK Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Got RTP packet from 87.238.72.149:17794 (type 03, seq 011680, ts 1266403830, len 000033) Sent RTP packet to 87.238.72.149:17794 (type 03, seq 025413, ts 000160, len 000033) Got RTP packet from 87.238.72.149:17794 (type 03, seq 011681, ts 1266403990, len 000033) Sent RTP packet to 87.238.72.149:17794 (type 03, seq 025414, ts 000320, len 000033) Got RTP packet from 87.238.72.149:17794 (type 03, seq 011682, ts 1266404150, len 000033) Sent RTP packet to 87.238.72.149:17794 (type 03, seq 025415, ts 000480, len 000033) Got RTP packet from 87.238.72.134:39526 (type 08, seq 000000, ts 000000, len 000160) <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK54b66f44;rport=5060 Record-Route: Record-Route: From: "07743898503" ;tag=as4a4d4d0e To: ;tag=BBaU28ee1U7yp Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 103 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=- 11706013 0 IN IP4 87.238.72.134 s=Cisco SDP 0 c=IN IP4 87.238.72.134 t=0 0 m=audio 39526 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.134:39526 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=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 87.238.72.134:39526 -- SIP/outbound-081bb338 is making progress passing it to Local/dialOffice@main-1fb8,2 -- Local/dialOffice@main-1fb8,1 is making progress passing it to SIP/87.238.72.149-081c29f0 Got RTP packet from 87.238.72.134:39526 (type 08, seq 000015, ts 002400, len 000160) Got RTP packet from 87.238.72.149:17794 (type 03, seq 011698, ts 1266406710, len 000033) Sent RTP packet to 87.238.72.149:17794 (type 03, seq 025431, ts 003040, len 000033) Got RTP packet from 87.238.72.134:39526 (type 08, seq 000016, ts 002560, len 000160) Got RTP packet from 87.238.72.149:17794 (type 03, seq 011699, ts 1266406870, len 000033) Sent RTP packet to 87.238.72.149:17794 (type 03, seq 025432, ts 003200, len 000033) Got RTP packet from 87.238.72.134:39526 (type 08, seq 000017, ts 002720, len 000160) <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK54b66f44;rport=5060 Record-Route: Record-Route: From: "07743898503" ;tag=as4a4d4d0e To: ;tag=BBaU28ee1U7yp Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 103 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 11706013 0 IN IP4 87.238.72.134 s=Cisco SDP 0 c=IN IP4 87.238.72.134 t=0 0 m=audio 39526 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (15 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.134:39526 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=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 87.238.72.134:39526 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:87.238.72.134 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK6f11a678;rport Route: , From: "07743898503" ;tag=as4a4d4d0e To: ;tag=BBaU28ee1U7yp Contact: Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/outbound-081bb338 answered Local/dialOffice@main-1fb8,2 Got RTP packet from 87.238.72.134:39526 (type 08, seq 000154, ts 024640, len 000160) -- Local/dialOffice@main-1fb8,1 answered SIP/87.238.72.149-081c29f0 Sent RTP packet to 87.238.72.149:17794 (type 03, seq 025570, ts 024640, len 000033) Got RTP packet from 87.238.72.149:17794 (type 03, seq 011837, ts 1266428950, len 000033) Sent RTP packet to 87.238.72.134:39526 (type 08, seq 014373, ts 1266428944, len 000160) Got RTP packet from 87.238.72.134:39526 (type 08, seq 000155, ts 024800, len 000160) -- Native bridging SIP/87.238.72.149-081c29f0 and SIP/outbound-081bb338 set_destination: Parsing for address/port to send to set_destination: set destination to 87.238.72.153, port 5060 Audio is at 213.232.93.17 port 17504 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 87.238.72.153:5060: INVITE sip:07743898503@87.238.72.149:5060 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK49a5bd00;rport Route: From: ;tag=as4e5059bd To: "07743898503" ;tag=39BEDB98-E11 Contact: Call-ID: E2451AB2-FDD911DD-A531F3EC-65114225@87.238.72.149 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 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: 242 v=0 o=root 17226 17227 IN IP4 87.238.72.134 s=session c=IN IP4 87.238.72.134 t=0 0 m=audio 39526 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 --- set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.135, port 5060 Audio is at 213.232.93.17 port 18068 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:87.238.72.134 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK565997ab;rport Route: , From: "07743898503" ;tag=as4a4d4d0e To: ;tag=BBaU28ee1U7yp Contact: Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 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: 242 v=0 o=root 17226 17228 IN IP4 87.238.72.149 s=session c=IN IP4 87.238.72.149 t=0 0 m=audio 17794 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 --- Got RTP packet from 87.238.72.149:17794 (type 03, seq 011838, ts 1266429110, len 000033) Sent RTP packet to 87.238.72.134:39526 (type 08, seq 014374, ts 1266429104, len 000160) == Spawn extension (main, dialOffice, 1) exited non-zero on 'Local/dialOffice@main-1fb8,2' <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK565997ab;rport=5060 From: "07743898503" ;tag=as4a4d4d0e To: ;tag=BBaU28ee1U7yp Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 104 INVITE Server: Sip EXpress router (0.8.99-dev (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Got RTP packet from 87.238.72.134:39526 (type 08, seq 000156, ts 024960, len 000160) Sent RTP packet to 87.238.72.149:17794 (type 03, seq 025571, ts 024960, len 000033) <--- SIP read from 87.238.72.153:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK49a5bd00;rport=5060 From: ;tag=as4e5059bd To: "07743898503" ;tag=39BEDB98-E11 Call-ID: E2451AB2-FDD911DD-A531F3EC-65114225@87.238.72.149 CSeq: 102 INVITE Server: OpenSER (1.2.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 87.238.72.153:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK49a5bd00;rport=5060 From: ;tag=as4e5059bd To: ;tag=39BEDB98-E11 Date: Thu, 19 Feb 2009 16:33:37 gmt Call-ID: E2451AB2-FDD911DD-A531F3EC-65114225@87.238.72.149 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 Remote-Party-ID: ;party=called;screen=yes;privacy=off Contact: Record-Route: Content-Type: application/sdp Content-Length: 249 v=0 o=CiscoSystemsSIP-GW-UserAgent 960 4613 IN IP4 87.238.72.149 s=SIP Call c=IN IP4 87.238.72.149 t=0 0 m=audio 17794 RTP/AVP 8 101 c=IN IP4 87.238.72.149 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 87.238.72.149:17794 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=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 87.238.72.149:17794 [Feb 19 16:33:37] DEBUG[17247]: chan_sip.c:5397 process_sdp: Oooh, we need to change our audio formats since our peer supports only 0x8 (alaw) and not 0x2 (gsm) set_destination: Parsing for address/port to send to set_destination: set destination to 87.238.72.153, port 5060 Transmitting (no NAT) to 87.238.72.153:5060: ACK sip:07743898503@87.238.72.149:5060 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK412d0ad8;rport Route: From: ;tag=as4e5059bd To: "07743898503" ;tag=39BEDB98-E11 Contact: Call-ID: E2451AB2-FDD911DD-A531F3EC-65114225@87.238.72.149 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK565997ab;rport=5060 Record-Route: Record-Route: From: "07743898503" ;tag=as4a4d4d0e To: ;tag=BBaU28ee1U7yp Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 104 INVITE Contact: User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Length: 181 v=0 o=- 11706013 1 IN IP4 87.238.72.134 s=Cisco SDP 0 c=IN IP4 87.238.72.134 t=0 0 m=audio 39526 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (15 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.134:39526 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=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 87.238.72.134:39526 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:87.238.72.134 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK0dfd6282;rport Route: , From: "07743898503" ;tag=as4a4d4d0e To: ;tag=BBaU28ee1U7yp Contact: Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- NATIVE BRIDGE WORKED *CLI> *CLI> *CLI> *CLI> HANGING UP CALL *CLI> *CLI> <--- SIP read from 87.238.72.153:5060 ---> BYE sip:08450045413@213.232.93.17:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKf33c.fe0d3b15.0 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bKB209BF1274 From: ;tag=39BEDB98-E11 To: ;tag=as4e5059bd Date: Thu, 19 Feb 2009 16:33:37 gmt Call-ID: E2451AB2-FDD911DD-A531F3EC-65114225@87.238.72.149 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 14 Timestamp: 1235061223 CSeq: 102 BYE Reason: Q.850;cause=16 Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 87.238.72.153 : 5060 (no NAT) <--- Transmitting (no NAT) to 87.238.72.153:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.238.72.153;branch=z9hG4bKf33c.fe0d3b15.0;received=87.238.72.153 Via: SIP/2.0/UDP 87.238.72.149:5060;branch=z9hG4bKB209BF1274 Record-Route: From: ;tag=39BEDB98-E11 To: ;tag=as4e5059bd Call-ID: E2451AB2-FDD911DD-A531F3EC-65114225@87.238.72.149 CSeq: 102 BYE User-Agent: Asterisk PBX 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 213.232.93.17 port 18068 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:87.238.72.134 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK078bf50c;rport Route: , From: "07743898503" ;tag=as4a4d4d0e To: ;tag=BBaU28ee1U7yp Contact: Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 105 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 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: 242 v=0 o=root 17226 17229 IN IP4 213.232.93.17 s=session c=IN IP4 213.232.93.17 t=0 0 m=audio 18068 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 '60071dae6ca84ae251d36f37549dc64e@213.232.93.17' in 32000 ms (Method: INVITE) == Spawn extension (main, 448450045413, 3) exited non-zero on 'SIP/87.238.72.149-081c29f0' <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK078bf50c;rport=5060 From: "07743898503" ;tag=as4a4d4d0e To: ;tag=BBaU28ee1U7yp Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 105 INVITE Server: Sip EXpress router (0.8.99-dev (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- 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:87.238.72.134 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK595444fc;rport Route: , From: "07743898503" ;tag=as4a4d4d0e To: ;tag=BBaU28ee1U7yp Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 106 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="ordersip", realm="gk.magrathea.net", algorithm=MD5, uri="sip:87.238.72.134", nonce="499d8b0ada57146340a9aa6d1ac6382fdd197079", response="ee587d9c7db92eb1e6e7ffa40efe8224", opaque="" Content-Length: 0 --- Scheduling destruction of SIP dialog '60071dae6ca84ae251d36f37549dc64e@213.232.93.17' in 32000 ms (Method: INVITE) Really destroying SIP dialog 'E2451AB2-FDD911DD-A531F3EC-65114225@87.238.72.149' Method: BYE <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK595444fc;rport=5060 From: "07743898503" ;tag=as4a4d4d0e To: ;tag=BBaU28ee1U7yp Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 106 BYE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 487 Session Terminated Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK078bf50c;rport=5060 From: "07743898503" ;tag=as4a4d4d0e To: ;tag=BBaU28ee1U7yp Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 105 INVITE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 487 Session Terminated Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK078bf50c;rport=5060 From: "07743898503" ;tag=as4a4d4d0e To: ;tag=BBaU28ee1U7yp Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 105 INVITE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (10 headers 0 lines) --- *CLI> *CLI> *CLI> <--- SIP read from 213.166.5.135:5060 ---> SIP/2.0 487 Session Terminated Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK078bf50c;rport=5060 From: "07743898503" ;tag=as4a4d4d0e To: ;tag=BBaU28ee1U7yp Call-ID: 60071dae6ca84ae251d36f37549dc64e@213.232.93.17 CSeq: 105 INVITE User-Agent: Avon v1.0 Allow: INVITE, CANCEL, BYE, ACK, PRACK Supported: timer, 100rel Content-Length: 0 <-------------> --- (10 headers 0 lines) --- *CLI> Beginning asterisk shutdown.... Executing last minute cleanups == Destroying musiconhold processes Asterisk cleanly ending (0). socrates:/usr/src/asterisk-1.4.13#