Asterisk 1.6.0.3 call to 0208 working reinvite (213 group gateway) <--- SIP read from UDP://213.166.5.130:5060 ---> INVITE sip:02080996227@213.232.93.17 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK3878.f3d091e6.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK16BDB034CA Remote-Party-ID: ;party=calling;screen=yes;privacy=off From: "07743898503" ;tag=463C1DC4-F6F To: Date: Thu, 19 Feb 2009 16:49:35 gmt Call-ID: 1F9301F7-FDDC11DD-83CD8556-8AABAA75@213.166.5.140 Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 14 Timestamp: 1235062175 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 1707 3330 IN IP4 213.166.5.140 s=SIP Call c=IN IP4 213.166.5.140 t=0 0 m=audio 20568 RTP/AVP 8 18 4 3 98 0 101 c=IN IP4 213.166.5.140 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 SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Sending to 213.166.5.130 : 5060 (no NAT) Using INVITE request as basis request - 1F9301F7-FDDC11DD-83CD8556-8AABAA75@213.166.5.140 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 213.166.5.140:20568 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 213.166.5.140:20568 Looking for 02080996227 in main (domain 213.232.93.17) 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=z9hG4bK3878.f3d091e6.0;received=213.166.5.130 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK16BDB034CA Record-Route: From: "07743898503" ;tag=463C1DC4-F6F To: Call-ID: 1F9301F7-FDDC11DD-83CD8556-8AABAA75@213.166.5.140 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 [02080996227@main:1] Answer("SIP/213.166.5.140-0824f890", "") in new stack Audio is at 213.232.93.17 port 14146 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=z9hG4bK3878.f3d091e6.0;received=213.166.5.130 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK16BDB034CA Record-Route: From: "07743898503" ;tag=463C1DC4-F6F To: ;tag=as33a2e94a Call-ID: 1F9301F7-FDDC11DD-83CD8556-8AABAA75@213.166.5.140 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: 263 v=0 o=root 828038861 828038861 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 14146 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://213.166.5.130:5060 ---> ACK sip:02080996227@213.232.93.17:5060 SIP/2.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK3878.f3d091e6.2 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK16BDB05FA From: ;tag=463C1DC4-F6F To: ;tag=as33a2e94a Date: Thu, 19 Feb 2009 16:49:35 gmt Call-ID: 1F9301F7-FDDC11DD-83CD8556-8AABAA75@213.166.5.140 Max-Forwards: 14 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Got RTP packet from 213.166.5.140:20568 (type 08, seq 004335, ts 1513791379, len 000160) -- Executing [02080996227@main:2] Dial("SIP/213.166.5.140-0824f890", "Local/dialOffice@main,,ro") in new stack -- Called dialOffice@main -- Executing [dialOffice@main:1] Dial("Local/dialOffice@main-63c3;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 19724 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=z9hG4bK30b863ca;rport Max-Forwards: 70 From: "07743898503" ;tag=as03f810b8 To: Contact: Call-ID: 18237ae70cdbb9d51a7efa1308021a51@213.232.93.17 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.3 Date: Thu, 19 Feb 2009 16:49:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 266968768 266968768 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 19724 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=z9hG4bK30b863ca;rport=5060 From: "07743898503" ;tag=as03f810b8 To: ;tag=a3e149d53f0faf6bf88ecc254282dfd9.7889 Call-ID: 18237ae70cdbb9d51a7efa1308021a51@213.232.93.17 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="gk.magrathea.net", nonce="499d8ecc8b39872289a2771c8daa824653eb8335" 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=z9hG4bK30b863ca;rport Max-Forwards: 70 From: "07743898503" ;tag=as03f810b8 To: ;tag=a3e149d53f0faf6bf88ecc254282dfd9.7889 Contact: Call-ID: 18237ae70cdbb9d51a7efa1308021a51@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 19724 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=z9hG4bK46e127d4;rport Max-Forwards: 70 From: "07743898503" ;tag=as03f810b8 To: Contact: Call-ID: 18237ae70cdbb9d51a7efa1308021a51@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="499d8ecc8b39872289a2771c8daa824653eb8335", response="c96b072a85c8bbcd8201c211a10fa3ad" Date: Thu, 19 Feb 2009 16:49:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 266968768 266968769 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 19724 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=z9hG4bK46e127d4;rport=5060 From: "07743898503" ;tag=as03f810b8 To: Call-ID: 18237ae70cdbb9d51a7efa1308021a51@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 213.166.5.140:20568 (type 08, seq 004336, ts 1513791539, len 000160) Sent RTP packet to 213.166.5.140:20568 (type 08, seq 006456, ts 000160, len 000160) Got RTP packet from 213.166.5.140:20568 (type 08, seq 004337, ts 1513791699, len 000160) Sent RTP packet to 213.166.5.140:20568 (type 08, seq 006457, ts 000320, len 000160) <--- 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=z9hG4bK46e127d4;rport=5060 Record-Route: Record-Route: From: "07743898503" ;tag=as03f810b8 To: ;tag=SpNt7a34pDmcr Call-ID: 18237ae70cdbb9d51a7efa1308021a51@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: 180 v=0 o=- 9262338 0 IN IP4 87.238.72.133 s=Cisco SDP 0 c=IN IP4 87.238.72.133 t=0 0 m=audio 49582 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 87.238.72.133:49582 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 87.238.72.133:49582 -- SIP/outbound-08255878 is making progress passing it to Local/dialOffice@main-63c3;2 -- Local/dialOffice@main-63c3;1 is making progress passing it to SIP/213.166.5.140-0824f890 Got RTP packet from 87.238.72.133:49582 (type 08, seq 000013, ts 002080, len 000160) Got RTP packet from 213.166.5.140:20568 (type 08, seq 004372, ts 1513797299, len 000160) Sent RTP packet to 213.166.5.140:20568 (type 08, seq 006492, ts 005920, len 000160) Got RTP packet from 87.238.72.133:49582 (type 08, seq 000014, ts 002240, len 000160) Got RTP packet from 213.166.5.140:20568 (type 08, seq 004373, ts 1513797459, len 000160) Sent RTP packet to 213.166.5.140:20568 (type 08, seq 006493, ts 006080, len 000160) Got RTP packet from 87.238.72.133:49582 (type 08, seq 000015, ts 002400, len 000160) Got RTP packet from 213.166.5.140:20568 (type 08, seq 004374, ts 1513797619, len 000160) Sent RTP packet to 213.166.5.140:20568 (type 08, seq 006494, ts 006240, len 000160) <--- 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=z9hG4bK46e127d4;rport=5060 Record-Route: Record-Route: From: "07743898503" ;tag=as03f810b8 To: ;tag=SpNt7a34pDmcr Call-ID: 18237ae70cdbb9d51a7efa1308021a51@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: 180 v=0 o=- 9262338 0 IN IP4 87.238.72.133 s=Cisco SDP 0 c=IN IP4 87.238.72.133 t=0 0 m=audio 49582 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=z9hG4bK739e7d22;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as03f810b8 To: ;tag=SpNt7a34pDmcr Contact: Call-ID: 18237ae70cdbb9d51a7efa1308021a51@213.232.93.17 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.0.3 Content-Length: 0 --- -- SIP/outbound-08255878 answered Local/dialOffice@main-63c3;2 -- Local/dialOffice@main-63c3;1 answered SIP/213.166.5.140-0824f890 Got RTP packet from 213.166.5.140:20568 (type 08, seq 004451, ts 1513809939, len 000160) Got RTP packet from 87.238.72.133:49582 (type 08, seq 000093, ts 014880, len 000160) -- Native bridging SIP/213.166.5.140-0824f890 and SIP/outbound-08255878 set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.130, port 5060 Audio is at 213.232.93.17 port 14146 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@213.166.5.140:5060 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK53b34035;rport Route: Max-Forwards: 70 From: ;tag=as33a2e94a To: "07743898503" ;tag=463C1DC4-F6F Contact: Call-ID: 1F9301F7-FDDC11DD-83CD8556-8AABAA75@213.166.5.140 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: 263 v=0 o=root 828038861 828038862 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 49582 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 19724 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=z9hG4bK5305b6e1;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as03f810b8 To: ;tag=SpNt7a34pDmcr Contact: Call-ID: 18237ae70cdbb9d51a7efa1308021a51@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: 263 v=0 o=root 266968768 266968770 IN IP4 213.166.5.140 s=Asterisk PBX 1.6.0.3 c=IN IP4 213.166.5.140 t=0 0 m=audio 20568 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://213.166.5.130:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK53b34035;rport=5060 From: ;tag=as33a2e94a To: "07743898503" ;tag=463C1DC4-F6F Call-ID: 1F9301F7-FDDC11DD-83CD8556-8AABAA75@213.166.5.140 CSeq: 102 INVITE Server: OpenSER (1.2.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- 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=z9hG4bK5305b6e1;rport=5060 From: "07743898503" ;tag=as03f810b8 To: ;tag=SpNt7a34pDmcr Call-ID: 18237ae70cdbb9d51a7efa1308021a51@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://213.166.5.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK53b34035;rport=5060 From: ;tag=as33a2e94a To: ;tag=463C1DC4-F6F Date: Thu, 19 Feb 2009 16:49:38 gmt Call-ID: 1F9301F7-FDDC11DD-83CD8556-8AABAA75@213.166.5.140 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 1707 3331 IN IP4 213.166.5.140 s=SIP Call c=IN IP4 213.166.5.140 t=0 0 m=audio 20568 RTP/AVP 8 101 c=IN IP4 213.166.5.140 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 213.166.5.140:20568 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 213.166.5.140:20568 set_destination: Parsing for address/port to send to set_destination: set destination to 213.166.5.130, port 5060 Transmitting (no NAT) to 213.166.5.130:5060: ACK sip:07743898503@213.166.5.140:5060 SIP/2.0 Via: SIP/2.0/UDP 213.232.93.17:5060;branch=z9hG4bK156548a4;rport Route: Max-Forwards: 70 From: ;tag=as33a2e94a To: "07743898503" ;tag=463C1DC4-F6F Contact: Call-ID: 1F9301F7-FDDC11DD-83CD8556-8AABAA75@213.166.5.140 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.3 Content-Length: 0 --- == Spawn extension (main, dialOffice, 1) exited non-zero on 'Local/dialOffice@main-63c3;2' <--- 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=z9hG4bK5305b6e1;rport=5060 Record-Route: Record-Route: From: "07743898503" ;tag=as03f810b8 To: ;tag=SpNt7a34pDmcr Call-ID: 18237ae70cdbb9d51a7efa1308021a51@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: 180 v=0 o=- 9262338 1 IN IP4 87.238.72.133 s=Cisco SDP 0 c=IN IP4 87.238.72.133 t=0 0 m=audio 49582 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 87.238.72.133:49582 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 87.238.72.133:49582 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=z9hG4bK0a272874;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as03f810b8 To: ;tag=SpNt7a34pDmcr Contact: Call-ID: 18237ae70cdbb9d51a7efa1308021a51@213.232.93.17 CSeq: 104 ACK User-Agent: Asterisk PBX 1.6.0.3 Content-Length: 0 --- *CLI> *CLI> REINVITE WORKED - FULL AUDIO *CLI> HANGING UP CALL *CLI> <--- SIP read from UDP://213.166.5.130:5060 ---> BYE sip:02080996227@213.232.93.17:5060 SIP/2.0 Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bK0878.c50517e1.0 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK16BDB69A90 From: ;tag=463C1DC4-F6F To: ;tag=as33a2e94a Date: Thu, 19 Feb 2009 16:49:38 gmt Call-ID: 1F9301F7-FDDC11DD-83CD8556-8AABAA75@213.166.5.140 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 14 Timestamp: 1235062184 CSeq: 102 BYE Content-Length: 0 <-------------> --- (12 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=z9hG4bK0878.c50517e1.0;received=213.166.5.130 Via: SIP/2.0/UDP 213.166.5.140:5060;branch=z9hG4bK16BDB69A90 From: ;tag=463C1DC4-F6F To: ;tag=as33a2e94a Call-ID: 1F9301F7-FDDC11DD-83CD8556-8AABAA75@213.166.5.140 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 19724 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=z9hG4bK48c128a9;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as03f810b8 To: ;tag=SpNt7a34pDmcr Contact: Call-ID: 18237ae70cdbb9d51a7efa1308021a51@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: 263 v=0 o=root 266968768 266968771 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 19724 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=z9hG4bK48c128a9;rport=5060 From: "07743898503" ;tag=as03f810b8 To: ;tag=SpNt7a34pDmcr Call-ID: 18237ae70cdbb9d51a7efa1308021a51@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 '18237ae70cdbb9d51a7efa1308021a51@213.232.93.17' in 32000 ms (Method: INVITE) == Spawn extension (main, 02080996227, 2) exited non-zero on 'SIP/213.166.5.140-0824f890' <--- 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=z9hG4bK48c128a9;rport=5060 Record-Route: Record-Route: From: "07743898503" ;tag=as03f810b8 To: ;tag=SpNt7a34pDmcr Call-ID: 18237ae70cdbb9d51a7efa1308021a51@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: 180 v=0 o=- 9262338 2 IN IP4 87.238.72.133 s=Cisco SDP 0 c=IN IP4 87.238.72.133 t=0 0 m=audio 49582 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 87.238.72.133:49582 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 87.238.72.133:49582 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=z9hG4bK0bc7e963;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as03f810b8 To: ;tag=SpNt7a34pDmcr Contact: Call-ID: 18237ae70cdbb9d51a7efa1308021a51@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=z9hG4bK2728a6ca;rport Route: , Max-Forwards: 70 From: "07743898503" ;tag=as03f810b8 To: ;tag=SpNt7a34pDmcr Call-ID: 18237ae70cdbb9d51a7efa1308021a51@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="499d8ecc8b39872289a2771c8daa824653eb8335", response="77b7939003e81ad6ca23fb445fbbf7d3" Content-Length: 0 --- Scheduling destruction of SIP dialog '18237ae70cdbb9d51a7efa1308021a51@213.232.93.17' in 32000 ms (Method: INVITE) Really destroying SIP dialog '1F9301F7-FDDC11DD-83CD8556-8AABAA75@213.166.5.140' 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=z9hG4bK2728a6ca;rport=5060 From: "07743898503" ;tag=as03f810b8 To: ;tag=SpNt7a34pDmcr Call-ID: 18237ae70cdbb9d51a7efa1308021a51@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 '18237ae70cdbb9d51a7efa1308021a51@213.232.93.17' Method: INVITE *CLI> *CLI> *CLI> Beginning asterisk shutdown.... Executing last minute cleanups == Destroying musiconhold processes Asterisk cleanly ending (0). socrates:/usr/src/asterisk-1.6.0.3.patched#