Asterisk 1.6.0.3 call to 0845 NOT WORKING (87 group gateway). *CLI> <--- SIP read from UDP://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=z9hG4bK8645.6be4f492.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK688947618 From: "07743898503" ;tag=7DDF1164-A72 To: Date: Thu, 19 Feb 2009 16:41:46 gmt Call-ID: 7C8E6A1-FDDB11DD-86EBE542-CAA4DB9B@87.238.72.155 Supported: 100rel,timer 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: 1235061706 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 404 v=0 o=CiscoSystemsSIP-GW-UserAgent 9676 3772 IN IP4 87.238.72.155 s=SIP Call c=IN IP4 87.238.72.155 t=0 0 m=audio 18800 RTP/AVP 8 18 4 3 2 0 101 c=IN IP4 87.238.72.155 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) --- == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Sending to 87.238.72.153 : 5060 (no NAT) Using INVITE request as basis request - 7C8E6A1-FDDB11DD-86EBE542-CAA4DB9B@87.238.72.155 No user '07743898503' in SIP users list No matching peer for '07743898503' from '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 87.238.72.155:18800 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 2 Found audio description format PCMU for ID 0 Found audio 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)/text=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 87.238.72.155:18800 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=z9hG4bK8645.6be4f492.0;received=87.238.72.153 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK688947618 Record-Route: From: "07743898503" ;tag=7DDF1164-A72 To: Call-ID: 7C8E6A1-FDDB11DD-86EBE542-CAA4DB9B@87.238.72.155 CSeq: 101 INVITE User-Agent: Asterisk PBX 1.6.0.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [08450045413@main:1] Goto("SIP/87.238.72.155-08247b38", "448450045413,1") in new stack -- Goto (main,448450045413,1) -- Executing [448450045413@main:1] Ringing("SIP/87.238.72.155-08247b38", "") 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=z9hG4bK8645.6be4f492.0;received=87.238.72.153 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK688947618 Record-Route: From: "07743898503" ;tag=7DDF1164-A72 To: ;tag=as2d0f15a6 Call-ID: 7C8E6A1-FDDB11DD-86EBE542-CAA4DB9B@87.238.72.155 CSeq: 101 INVITE User-Agent: Asterisk PBX 1.6.0.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [448450045413@main:2] Answer("SIP/87.238.72.155-08247b38", "") in new stack Audio is at 213.232.93.17 port 15682 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=z9hG4bK8645.6be4f492.0;received=87.238.72.153 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK688947618 Record-Route: From: "07743898503" ;tag=7DDF1164-A72 To: ;tag=as2d0f15a6 Call-ID: 7C8E6A1-FDDB11DD-86EBE542-CAA4DB9B@87.238.72.155 CSeq: 101 INVITE User-Agent: Asterisk PBX 1.6.0.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 310 v=0 o=root 325463474 325463474 IN IP4 213.232.93.17 s=Asterisk PBX 1.6.0.3 c=IN IP4 213.232.93.17 t=0 0 m=audio 15682 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 <------------> <--- SIP read from UDP://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=z9hG4bK8645.6be4f492.2 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK688948E2B From: ;tag=7DDF1164-A72 To: ;tag=as2d0f15a6 Date: Thu, 19 Feb 2009 16:41:46 gmt Call-ID: 7C8E6A1-FDDB11DD-86EBE542-CAA4DB9B@87.238.72.155 Max-Forwards: 14 CSeq: 101 ACK Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Got RTP packet from 87.238.72.155:18800 (type 03, seq 012686, ts 677199952, len 000033) -- Executing [448450045413@main:3] Dial("SIP/87.238.72.155-08247b38", "Local/dialOffice@main,,ro") in new stack -- Called dialOffice@main -- Executing [dialOffice@main:1] Dial("Local/dialOffice@main-3106;2", "SIP/01392213713@outbound") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 213.232.93.17 port 15494 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=z9hG4bK6a7093dc;rport Max-Forwards: 70 From: "07743898503" ;tag=as2754b43e To: Contact: Call-ID: 589572c008fa9fa23987e798114ac650@213.232.93.17 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.3 Date: Thu, 19 Feb 2009 16:41:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 1236298606 1236298606 IN IP4 213.232.93.17 s=Asterisk PBX 1.6.0.3 c=IN IP4 213.232.93.17 t=0 0 m=audio 15494 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 UDP://213.166.5.135:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK6a7093dc;rport=5060 From: "07743898503" ;tag=as2754b43e To: ;tag=a3e149d53f0faf6bf88ecc254282dfd9.9d03 Call-ID: 589572c008fa9fa23987e798114ac650@213.232.93.17 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="gk.magrathea.net", nonce="499d8cf77c3bf2ce0b672a7daf3b632935211c39" 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=z9hG4bK6a7093dc;rport Max-Forwards: 70 From: "07743898503" ;tag=as2754b43e To: ;tag=a3e149d53f0faf6bf88ecc254282dfd9.9d03 Contact: Call-ID: 589572c008fa9fa23987e798114ac650@213.232.93.17 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.3 Content-Length: 0 --- Audio is at 213.232.93.17 port 15494 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=z9hG4bK6914dc65;rport Max-Forwards: 70 From: "07743898503" ;tag=as2754b43e To: Contact: Call-ID: 589572c008fa9fa23987e798114ac650@213.232.93.17 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.0.3 Proxy-Authorization: Digest username="ordersip", realm="gk.magrathea.net", algorithm=MD5, uri="sip:01392213713@gk.magrathea.net", nonce="499d8cf77c3bf2ce0b672a7daf3b632935211c39", response="d5ec468b241395668703e448a67389a1" Date: Thu, 19 Feb 2009 16:41:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 1236298606 1236298607 IN IP4 213.232.93.17 s=Asterisk PBX 1.6.0.3 c=IN IP4 213.232.93.17 t=0 0 m=audio 15494 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 UDP://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=z9hG4bK6914dc65;rport=5060 From: "07743898503" ;tag=as2754b43e To: Call-ID: 589572c008fa9fa23987e798114ac650@213.232.93.17 CSeq: 103 INVITE Server: Sip EXpress router (0.8.99-dev (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Got RTP packet from 87.238.72.155:18800 (type 03, seq 012687, ts 677200112, len 000033) Sent RTP packet to 87.238.72.155:18800 (type 03, seq 020827, ts 000160, len 000033) Got RTP packet from 87.238.72.155:18800 (type 03, seq 012688, ts 677200272, len 000033) Sent RTP packet to 87.238.72.155:18800 (type 03, seq 020828, ts 000320, len 000033) Got RTP packet from 87.238.72.155:18800 (type 03, seq 012689, ts 677200432, len 000033) Sent RTP packet to 87.238.72.155:18800 (type 03, seq 020829, ts 000480, len 000033) Got RTP packet from 87.238.72.155:18800 (type 03, seq 012690, ts 677200592, len 000033) Sent RTP packet to 87.238.72.155:18800 (type 03, seq 020830, ts 000640, len 000033) <--- SIP read from UDP://213.166.5.135:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK6914dc65;rport=5060 Record-Route: Record-Route: From: "07743898503" ;tag=as2754b43e To: ;tag=Qp1BXH10rSKmr Call-ID: 589572c008fa9fa23987e798114ac650@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=- 13508132 0 IN IP4 87.238.72.133 s=Cisco SDP 0 c=IN IP4 87.238.72.133 t=0 0 m=audio 44834 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 87.238.72.133:44834 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 87.238.72.133:44834 -- SIP/outbound-08263c28 is making progress passing it to Local/dialOffice@main-3106;2 -- Local/dialOffice@main-3106;1 is making progress passing it to SIP/87.238.72.155-08247b38 Got RTP packet from 87.238.72.133:44834 (type 08, seq 000009, ts 001440, len 000160) Got RTP packet from 87.238.72.155:18800 (type 03, seq 012721, ts 677205552, len 000033) Sent RTP packet to 87.238.72.155:18800 (type 03, seq 020861, ts 005600, len 000033) Got RTP packet from 87.238.72.133:44834 (type 08, seq 000010, ts 001600, len 000160) Got RTP packet from 87.238.72.155:18800 (type 03, seq 012722, ts 677205712, len 000033) Sent RTP packet to 87.238.72.155:18800 (type 03, seq 020862, ts 005760, len 000033) Got RTP packet from 87.238.72.133:44834 (type 08, seq 000011, ts 001760, len 000160) Got RTP packet from 87.238.72.155:18800 (type 03, seq 012723, ts 677205872, len 000033) Sent RTP packet to 87.238.72.155:18800 (type 03, seq 020863, ts 005920, len 000033) Got RTP packet from 87.238.72.133:44834 (type 08, seq 000012, ts 001920, len 000160) Got RTP packet from 87.238.72.155:18800 (type 03, seq 012724, ts 677206032, len 000033) Sent RTP packet to 87.238.72.155:18800 (type 03, seq 020864, ts 006080, len 000033) Got RTP packet from 87.238.72.133:44834 (type 08, seq 000013, ts 002080, len 000160) Got RTP packet from 87.238.72.155:18800 (type 03, seq 012725, ts 677206192, len 000033) Sent RTP packet to 87.238.72.155:18800 (type 03, seq 020865, ts 006240, len 000033) Got RTP packet from 87.238.72.133:44834 (type 08, seq 000014, ts 002240, len 000160) Got RTP packet from 87.238.72.155:18800 (type 03, seq 012726, ts 677206352, len 000033) Sent RTP packet to 87.238.72.155:18800 (type 03, seq 020866, ts 006400, len 000033) Got RTP packet from 87.238.72.133:44834 (type 08, seq 000015, ts 002400, len 000160) Got RTP packet from 87.238.72.155:18800 (type 03, seq 012727, ts 677206512, len 000033) Sent RTP packet to 87.238.72.155:18800 (type 03, seq 020867, ts 006560, len 000033) <--- SIP read from UDP://213.166.5.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK6914dc65;rport=5060 Record-Route: Record-Route: From: "07743898503" ;tag=as2754b43e To: ;tag=Qp1BXH10rSKmr Call-ID: 589572c008fa9fa23987e798114ac650@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=- 13508132 0 IN IP4 87.238.72.133 s=Cisco SDP 0 c=IN IP4 87.238.72.133 t=0 0 m=audio 44834 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (15 headers 9 lines) --- 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:87.238.72.133 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK747856ce;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as2754b43e To: ;tag=Qp1BXH10rSKmr Contact: Call-ID: 589572c008fa9fa23987e798114ac650@213.232.93.17 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.0.3 Content-Length: 0 --- Got RTP packet from 87.238.72.155:18800 (type 03, seq 012839, ts 677224432, len 000033) Sent RTP packet to 87.238.72.155:18800 (type 03, seq 020979, ts 024480, len 000033) -- SIP/outbound-08263c28 answered Local/dialOffice@main-3106;2 -- Local/dialOffice@main-3106;1 answered SIP/87.238.72.155-08247b38 Got RTP packet from 87.238.72.155:18800 (type 03, seq 012840, ts 677224592, len 000033) Got RTP packet from 87.238.72.133:44834 (type 08, seq 000128, ts 020480, len 000160) Sent RTP packet to 87.238.72.155:18800 (type 03, seq 020980, ts 020480, len 000033) Got RTP packet from 87.238.72.133:44834 (type 08, seq 000129, ts 020640, len 000160) -- Native bridging SIP/87.238.72.155-08247b38 and SIP/outbound-08263c28 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 15682 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: INVITE sip:07743898503@87.238.72.155:5060 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK5db6285d;rport Route: Max-Forwards: 70 From: ;tag=as2d0f15a6 To: "07743898503" ;tag=7DDF1164-A72 Contact: Call-ID: 7C8E6A1-FDDB11DD-86EBE542-CAA4DB9B@87.238.72.155 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 310 v=0 o=root 325463474 325463475 IN IP4 87.238.72.133 s=Asterisk PBX 1.6.0.3 c=IN IP4 87.238.72.133 t=0 0 m=audio 44834 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 --- 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 15494 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:87.238.72.133 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK136c3be9;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as2754b43e To: ;tag=Qp1BXH10rSKmr Contact: Call-ID: 589572c008fa9fa23987e798114ac650@213.232.93.17 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.6.0.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 265 v=0 o=root 1236298606 1236298608 IN IP4 87.238.72.155 s=Asterisk PBX 1.6.0.3 c=IN IP4 87.238.72.155 t=0 0 m=audio 18800 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 87.238.72.155:18800 (type 03, seq 012841, ts 677224752, len 000033) Sent RTP packet to 87.238.72.133:44834 (type 08, seq 015104, ts 677224752, len 000160) <--- SIP read from UDP://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=z9hG4bK136c3be9;rport=5060 From: "07743898503" ;tag=as2754b43e To: ;tag=Qp1BXH10rSKmr Call-ID: 589572c008fa9fa23987e798114ac650@213.232.93.17 CSeq: 104 INVITE Server: Sip EXpress router (0.8.99-dev (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP://87.238.72.153:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK5db6285d;rport=5060 From: ;tag=as2d0f15a6 To: "07743898503" ;tag=7DDF1164-A72 Call-ID: 7C8E6A1-FDDB11DD-86EBE542-CAA4DB9B@87.238.72.155 CSeq: 102 INVITE Server: OpenSER (1.2.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- == Spawn extension (main, dialOffice, 1) exited non-zero on 'Local/dialOffice@main-3106;2' <--- SIP read from UDP://87.238.72.153:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK5db6285d;rport=5060 From: ;tag=as2d0f15a6 To: ;tag=7DDF1164-A72 Date: Thu, 19 Feb 2009 16:41:49 gmt Call-ID: 7C8E6A1-FDDB11DD-86EBE542-CAA4DB9B@87.238.72.155 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 9676 3773 IN IP4 87.238.72.155 s=SIP Call c=IN IP4 87.238.72.155 t=0 0 m=audio 18800 RTP/AVP 3 101 c=IN IP4 87.238.72.155 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 <-------------> --- (15 headers 11 lines) --- Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 87.238.72.155:18800 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 87.238.72.155:18800 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@87.238.72.155:5060 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK0efe901c;rport Route: Max-Forwards: 70 From: ;tag=as2d0f15a6 To: "07743898503" ;tag=7DDF1164-A72 Contact: Call-ID: 7C8E6A1-FDDB11DD-86EBE542-CAA4DB9B@87.238.72.155 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.3 Content-Length: 0 --- <--- SIP read from UDP://213.166.5.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK136c3be9;rport=5060 Record-Route: Record-Route: From: "07743898503" ;tag=as2754b43e To: ;tag=Qp1BXH10rSKmr Call-ID: 589572c008fa9fa23987e798114ac650@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=- 13508132 1 IN IP4 87.238.72.133 s=Cisco SDP 0 c=IN IP4 87.238.72.133 t=0 0 m=audio 44834 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 87.238.72.133:44834 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 87.238.72.133:44834 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:87.238.72.133 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK50eac24a;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as2754b43e To: ;tag=Qp1BXH10rSKmr Contact: Call-ID: 589572c008fa9fa23987e798114ac650@213.232.93.17 CSeq: 104 ACK User-Agent: Asterisk PBX 1.6.0.3 Content-Length: 0 --- *CLI> *CLI> *CLI> REINVITE NOT WORKING - NO AUDIO *CLI> *CLI> HANGING UP CALL *CLI> <--- SIP read from UDP://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=z9hG4bK5645.f3e1aa44.0 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK6889501D03 From: ;tag=7DDF1164-A72 To: ;tag=as2d0f15a6 Date: Thu, 19 Feb 2009 16:41:49 gmt Call-ID: 7C8E6A1-FDDB11DD-86EBE542-CAA4DB9B@87.238.72.155 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 14 Timestamp: 1235061716 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=z9hG4bK5645.f3e1aa44.0;received=87.238.72.153 Via: SIP/2.0/UDP 87.238.72.155:5060;branch=z9hG4bK6889501D03 Record-Route: From: ;tag=7DDF1164-A72 To: ;tag=as2d0f15a6 Call-ID: 7C8E6A1-FDDB11DD-86EBE542-CAA4DB9B@87.238.72.155 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.0.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer 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 15494 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:87.238.72.133 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK645dbc39;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as2754b43e To: ;tag=Qp1BXH10rSKmr Contact: Call-ID: 589572c008fa9fa23987e798114ac650@213.232.93.17 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.6.0.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 265 v=0 o=root 1236298606 1236298609 IN IP4 213.232.93.17 s=Asterisk PBX 1.6.0.3 c=IN IP4 213.232.93.17 t=0 0 m=audio 15494 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 UDP://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=z9hG4bK645dbc39;rport=5060 From: "07743898503" ;tag=as2754b43e To: ;tag=Qp1BXH10rSKmr Call-ID: 589572c008fa9fa23987e798114ac650@213.232.93.17 CSeq: 105 INVITE Server: Sip EXpress router (0.8.99-dev (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Scheduling destruction of SIP dialog '589572c008fa9fa23987e798114ac650@213.232.93.17' in 32000 ms (Method: INVITE) == Spawn extension (main, 448450045413, 3) exited non-zero on 'SIP/87.238.72.155-08247b38' <--- SIP read from UDP://213.166.5.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK645dbc39;rport=5060 Record-Route: Record-Route: From: "07743898503" ;tag=as2754b43e To: ;tag=Qp1BXH10rSKmr Call-ID: 589572c008fa9fa23987e798114ac650@213.232.93.17 CSeq: 105 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=- 13508132 2 IN IP4 87.238.72.133 s=Cisco SDP 0 c=IN IP4 87.238.72.133 t=0 0 m=audio 44834 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 87.238.72.133:44834 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 87.238.72.133:44834 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:87.238.72.133 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK537173eb;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as2754b43e To: ;tag=Qp1BXH10rSKmr Contact: Call-ID: 589572c008fa9fa23987e798114ac650@213.232.93.17 CSeq: 105 ACK User-Agent: Asterisk PBX 1.6.0.3 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:87.238.72.133 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK086ed4fd;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as2754b43e To: ;tag=Qp1BXH10rSKmr Call-ID: 589572c008fa9fa23987e798114ac650@213.232.93.17 CSeq: 106 BYE User-Agent: Asterisk PBX 1.6.0.3 Proxy-Authorization: Digest username="ordersip", realm="gk.magrathea.net", algorithm=MD5, uri="sip:87.238.72.133", nonce="499d8cf77c3bf2ce0b672a7daf3b632935211c39", response="08feaf518c595b916149bece6c931e28" Content-Length: 0 --- Scheduling destruction of SIP dialog '589572c008fa9fa23987e798114ac650@213.232.93.17' in 32000 ms (Method: INVITE) Really destroying SIP dialog '7C8E6A1-FDDB11DD-86EBE542-CAA4DB9B@87.238.72.155' Method: BYE <--- SIP read from UDP://213.166.5.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK086ed4fd;rport=5060 From: "07743898503" ;tag=as2754b43e To: ;tag=Qp1BXH10rSKmr Call-ID: 589572c008fa9fa23987e798114ac650@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) --- Really destroying SIP dialog '589572c008fa9fa23987e798114ac650@213.232.93.17' Method: INVITE *CLI> *CLI> *CLI> *CLI>