=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2011.03.04 09:56:06 =~=~=~=~=~=~=~=~=~=~=~= voice_switch_0*CLI> [0KSIP Debugging enabled [Kvoice_switch_0*CLI> voice_switch_0*CLI> core set debug on voice_switch_0*CLI> [0KUsage: core set {debug|verbose} [atleast] [filename] core set {debug|verbose} off Sets level of debug or verbose messages to be displayed or sets a filename to display debug messages from. 0 or off means no messages should be displayed. Equivalent to -d[d[...]] or -v[v[v...]] on startup [Kvoice_switch_0*CLI> core set debug on[K[K9 voice_switch_0*CLI> [0KCore debug is at least 9 [Kvoice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.9:5060 ---> INVITE sip:+74956986015@192.168.0.28:5060;user=phone SIP/2.0 From: 79166612794 ;tag=8036dfa8-0-13c4-45026-1e900-7816b79f-1e900 To: Call-ID: 5F92B305-5D6E-4F93-84B1-B1DCF498E05A CSeq: 1 INVITE Via: SIP/2.0/UDP 82.204.187.9:5060;branch=z9hG4bK-1e900-7762914-127be005 P-Asserted-Identity: X-VT-Corr: 4003={5F92B305-5D6E-4F93-84B1-B1DCF498E05A},30=82.204.187.9,31=Comtelco_CX X-VT-Call: 4001=SH1/BR1/TR1@82.204.187.7 P-Charging-Vector: icid-value=5F92B305-5D6E-4F93-84B1-B1DCF498E05A; icid-generated-at=82.204.187.9 Supported: 100rel Allow: INVITE,ACK,BYE,CANCEL,INFO,PRACK,OPTIONS Max-Forwards: 70 Contact: User-Agent: vocl-essentra-cx/8.0F2-19020-48 Content-Type: application/sdp Content-Length: 289 v=0 o=Essentra-CX-AC 3175325 1 IN IP4 82.204.187.9 s=- c=IN IP4 82.204.187.7 t=0 0 m=audio 4140 RTP/AVP 18 4 8 0 96 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000/1 a=fmtp:4 bitrate=6.3;annexa=yes a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=silenceSupp:off - - - - <-------------> [Kvoice_switch_0*CLI> [0K--- (17 headers 13 lines) --- [Kvoice_switch_0*CLI> [0K == Using SIP RTP CoS mark 5 [Kvoice_switch_0*CLI> [0K == Using UDPTL CoS mark 5 [Kvoice_switch_0*CLI> [0KSending to 82.204.187.9 : 5060 (no NAT) [Kvoice_switch_0*CLI> [0KUsing INVITE request as basis request - 5F92B305-5D6E-4F93-84B1-B1DCF498E05A Found peer 'CX' for '+79166612794' from 82.204.187.9:5060 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 96 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 96 Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 82.204.187.7:4140 Looking for +74956986015 in fromCX (domain 192.168.0.28) list_route: hop: <--- Transmitting (no NAT) to 82.204.187.9:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 82.204.187.9:5060;branch=z9hG4bK-1e900-7762914-127be005;received=82.204.187.9 From: 79166612794 ;tag=8036dfa8-0-13c4-45026-1e900-7816b79f-1e900 To: Call-ID: 5F92B305-5D6E-4F93-84B1-B1DCF498E05A CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Kvoice_switch_0*CLI> [0K -- Executing [+74956986015@fromCX:1] [1;36mAnswer[0m("[1;35mSIP/CX-0000012e[0m", "[1;35m[0m") in new stack [Kvoice_switch_0*CLI> [0KAudio is at 192.168.0.28 port 19136 [Kvoice_switch_0*CLI> [0KAdding codec 0x100 (g729) to SDP [Kvoice_switch_0*CLI> [0KAdding codec 0x4 (ulaw) to SDP [Kvoice_switch_0*CLI> [0KAdding codec 0x8 (alaw) to SDP [Kvoice_switch_0*CLI> [0KAdding non-codec 0x1 (telephone-event) to SDP [Kvoice_switch_0*CLI> [0K <--- Reliably Transmitting (no NAT) to 82.204.187.9:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.204.187.9:5060;branch=z9hG4bK-1e900-7762914-127be005;received=82.204.187.9 From: 79166612794 ;tag=8036dfa8-0-13c4-45026-1e900-7816b79f-1e900 To: ;tag=as03652b84 Call-ID: 5F92B305-5D6E-4F93-84B1-B1DCF498E05A CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 332 v=0 o=root 1510061740 1510061740 IN IP4 192.168.0.28 s=Asterisk PBX 1.6.2.17 c=IN IP4 192.168.0.28 t=0 0 m=audio 19136 RTP/AVP 18 0 8 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.9:5060 ---> ACK sip:+74956986015@192.168.0.28 SIP/2.0 From: 79166612794 ;tag=8036dfa8-0-13c4-45026-1e900-7816b79f-1e900 To: ;tag=as03652b84 Call-ID: 5F92B305-5D6E-4F93-84B1-B1DCF498E05A CSeq: 1 ACK Via: SIP/2.0/UDP 82.204.187.9:5060;branch=z9hG4bK-1e900-7762926-4089e02d Max-Forwards: 70 User-Agent: vocl-essentra-cx/8.0F2-19020-48 Contact: Content-Length: 0 <-------------> [Kvoice_switch_0*CLI> [0K--- (10 headers 0 lines) --- [Kvoice_switch_0*CLI> [0K -- Executing [+74956986015@fromCX:2] [1;36mWait[0m("[1;35mSIP/CX-0000012e[0m", "[1;35m1[0m") in new stack [Kvoice_switch_0*CLI> [0K -- Executing [+74956986015@fromCX:3] [1;36mDial[0m("[1;35mSIP/CX-0000012e[0m", "[1;35mSIP/BAX/74956986015[0m") in new stack == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Audio is at 192.168.0.28 port 10130 Adding codec 0x100 (g729) to SDP 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 82.204.187.10:5060: INVITE sip:74956986015@82.204.187.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK5eaf0408;rport Max-Forwards: 70 From: "79166612794" ;tag=as7bfdce7a To: Contact: Call-ID: 2bb2cc33494ef8e60564e4fc17525f65@192.168.0.28 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.17 Date: Fri, 04 Mar 2011 06:57:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 355 v=0 o=root 1912272497 1912272497 IN IP4 192.168.0.28 s=Asterisk PBX 1.6.2.17 c=IN IP4 192.168.0.28 t=0 0 m=audio 10130 RTP/AVP 18 3 0 8 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called BAX/74956986015 [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> SIP/2.0 100 Trying From: "79166612794";tag=as7bfdce7a To: ;tag=c317be0 Call-ID: 2bb2cc33494ef8e60564e4fc17525f65@192.168.0.28 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK5eaf0408 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> SIP/2.0 183 Session Progress From: "79166612794";tag=as7bfdce7a To: ;tag=c317be0 Call-ID: 2bb2cc33494ef8e60564e4fc17525f65@192.168.0.28 CSeq: 102 INVITE Server: vocl-essentra-bax/8.0.251 Contact: Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK5eaf0408 Content-Type: application/sdp Content-Length: 206 v=0 o=Essentra-Relay 3508210658 3508210658 IN IP4 82.204.187.10 s=IVR c=IN IP4 82.204.187.10 t=0 0 m=audio 41810 RTP/AVP 18 96 a=rtpmap:18 g729/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> --- (10 headers 9 lines) --- Found RTP audio format 18 Found RTP audio format 96 Found audio description format g729 for ID 18 Found audio description format telephone-event for ID 96 Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 82.204.187.10:41810 [Kvoice_switch_0*CLI> [0K -- SIP/BAX-0000012f is making progress passing it to SIP/CX-0000012e [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.9:5060 ---> BYE sip:+74956986015@192.168.0.28 SIP/2.0 From: 79166612794 ;tag=8036dfa8-0-13c4-45026-1e900-7816b79f-1e900 To: ;tag=as03652b84 Call-ID: 5F92B305-5D6E-4F93-84B1-B1DCF498E05A CSeq: 2 BYE Via: SIP/2.0/UDP 82.204.187.9:5060;branch=z9hG4bK-1e907-7764484-47ea958c Reason: Q.850 ;cause=16;text="Normal call clearing" Max-Forwards: 70 User-Agent: vocl-essentra-cx/8.0F2-19020-48 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 82.204.187.9 : 5060 (no NAT) <--- Transmitting (no NAT) to 82.204.187.9:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.204.187.9:5060;branch=z9hG4bK-1e907-7764484-47ea958c;received=82.204.187.9 From: 79166612794 ;tag=8036dfa8-0-13c4-45026-1e900-7816b79f-1e900 To: ;tag=as03652b84 Call-ID: 5F92B305-5D6E-4F93-84B1-B1DCF498E05A CSeq: 2 BYE Server: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [Kvoice_switch_0*CLI> [0KScheduling destruction of SIP dialog '2bb2cc33494ef8e60564e4fc17525f65@192.168.0.28' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 82.204.187.10:5060: CANCEL sip:74956986015@82.204.187.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK5eaf0408;rport Max-Forwards: 70 From: "79166612794" ;tag=as7bfdce7a To: Call-ID: 2bb2cc33494ef8e60564e4fc17525f65@192.168.0.28 CSeq: 102 CANCEL User-Agent: Asterisk PBX 1.6.2.17 Content-Length: 0 --- Scheduling destruction of SIP dialog '2bb2cc33494ef8e60564e4fc17525f65@192.168.0.28' in 32000 ms (Method: INVITE) == Spawn extension (fromCX, +74956986015, 3) exited non-zero on 'SIP/CX-0000012e' [Mar 4 09:57:46] [0;31mERROR[0m[29006]: [1;37mcdr_sqlite3_custom.c[0m:[1;37m269[0m [1;37mwrite_cdr[0m: unable to open database file. SQL: INSERT INTO cdr (calldate,clid,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,test) VALUES ('2011-03-04 09:57:39','"79166612794" <+79166612794>','fromCX','SIP/CX-0000012e','SIP/BAX-0000012f','Dial','SIP/BAX/74956986015','7','7','NO ANSWER','DOCUMENTATION','','1299221859.302','',''). [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> SIP/2.0 200 OK From: "79166612794";tag=as7bfdce7a To: ;tag=c317be0 Call-ID: 2bb2cc33494ef8e60564e4fc17525f65@192.168.0.28 CSeq: 102 CANCEL Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK5eaf0408 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '5F92B305-5D6E-4F93-84B1-B1DCF498E05A' Method: BYE [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> SIP/2.0 487 Request Terminated From: "79166612794";tag=as7bfdce7a To: ;tag=c317be0 Call-ID: 2bb2cc33494ef8e60564e4fc17525f65@192.168.0.28 CSeq: 102 INVITE Server: vocl-essentra-bax/8.0.251 Contact: Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK5eaf0408 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Kvoice_switch_0*CLI> [0KTransmitting (no NAT) to 82.204.187.10:5060: ACK sip:74956986015@82.204.187.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK5eaf0408;rport Max-Forwards: 70 From: "79166612794" ;tag=as7bfdce7a To: ;tag=c317be0 Contact: Call-ID: 2bb2cc33494ef8e60564e4fc17525f65@192.168.0.28 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.17 Content-Length: 0 --- [Kvoice_switch_0*CLI> [0KReally destroying SIP dialog '2bb2cc33494ef8e60564e4fc17525f65@192.168.0.28' Method: INVITE [Kvoice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.9:5060 ---> INVITE sip:+74956986015@192.168.0.28:5060;user=phone SIP/2.0 From: 79166612794 ;tag=80370848-0-13c4-45026-1e943-3a77dd38-1e943 To: Call-ID: 881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A CSeq: 1 INVITE Via: SIP/2.0/UDP 82.204.187.9:5060;branch=z9hG4bK-1e943-777316a-5ca3802a P-Asserted-Identity: X-VT-Corr: 4003={881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A},30=82.204.187.9,31=Comtelco_CX X-VT-Call: 4001=SH1/BR1/TR1@82.204.187.7 P-Charging-Vector: icid-value=881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A; icid-generated-at=82.204.187.9 Supported: 100rel Allow: INVITE,ACK,BYE,CANCEL,INFO,PRACK,OPTIONS Max-Forwards: 70 Contact: User-Agent: vocl-essentra-cx/8.0F2-19020-48 Content-Type: application/sdp Content-Length: 292 v=0 o=Essentra-CX-AC 3013260210 1 IN IP4 82.204.187.9 s=- c=IN IP4 82.204.187.7 t=0 0 m=audio 4220 RTP/AVP 18 4 8 0 96 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000/1 a=fmtp:4 bitrate=6.3;annexa=yes a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=silenceSupp:off - - - - <-------------> --- (17 headers 13 lines) --- == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Sending to 82.204.187.9 : 5060 (no NAT) Using INVITE request as basis request - 881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A Found peer 'CX' for '+79166612794' from 82.204.187.9:5060 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 96 Found audio description format G723 for ID 4 [Kvoice_switch_0*CLI> [0KFound audio description format telephone-event for ID 96 Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 82.204.187.7:4220 Looking for +74956986015 in fromCX (domain 192.168.0.28) list_route: hop: <--- Transmitting (no NAT) to 82.204.187.9:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 82.204.187.9:5060;branch=z9hG4bK-1e943-777316a-5ca3802a;received=82.204.187.9 From: 79166612794 ;tag=80370848-0-13c4-45026-1e943-3a77dd38-1e943 To: Call-ID: 881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [+74956986015@fromCX:1] [1;36mAnswer[0m("[1;35mSIP/CX-00000130[0m", "[1;35m[0m") in new stack Audio is at 192.168.0.28 port 19238 Adding codec 0x100 (g729) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP [Kvoice_switch_0*CLI> [0KAdding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 82.204.187.9:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.204.187.9:5060;branch=z9hG4bK-1e943-777316a-5ca3802a;received=82.204.187.9 From: 79166612794 ;tag=80370848-0-13c4-45026-1e943-3a77dd38-1e943 To: ;tag=as4db5349f Call-ID: 881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 330 v=0 o=root 236836765 236836765 IN IP4 192.168.0.28 s=Asterisk PBX 1.6.2.17 c=IN IP4 192.168.0.28 t=0 0 m=audio 19238 RTP/AVP 18 0 8 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.9:5060 ---> ACK sip:+74956986015@192.168.0.28 SIP/2.0 From: 79166612794 ;tag=80370848-0-13c4-45026-1e943-3a77dd38-1e943 To: ;tag=as4db5349f Call-ID: 881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A CSeq: 1 ACK Via: SIP/2.0/UDP 82.204.187.9:5060;branch=z9hG4bK-1e943-777317b-7a3a4656 Max-Forwards: 70 User-Agent: vocl-essentra-cx/8.0F2-19020-48 Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Kvoice_switch_0*CLI> [0K -- Executing [+74956986015@fromCX:2] [1;36mWait[0m("[1;35mSIP/CX-00000130[0m", "[1;35m1[0m") in new stack [Kvoice_switch_0*CLI> [0K -- Executing [+74956986015@fromCX:3] [1;36mDial[0m("[1;35mSIP/CX-00000130[0m", "[1;35mSIP/BAX/74956986015[0m") in new stack == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Audio is at 192.168.0.28 port 18636 Adding codec 0x100 (g729) to SDP 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 82.204.187.10:5060: INVITE sip:74956986015@82.204.187.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK2387e191;rport Max-Forwards: 70 From: "79166612794" ;tag=as790ff91d To: Contact: Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.17 Date: Fri, 04 Mar 2011 06:58:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 353 v=0 o=root 727601229 727601229 IN IP4 192.168.0.28 s=Asterisk PBX 1.6.2.17 c=IN IP4 192.168.0.28 t=0 0 m=audio 18636 RTP/AVP 18 3 0 8 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called BAX/74956986015 [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> SIP/2.0 100 Trying From: "79166612794";tag=as790ff91d To: ;tag=c1cb190 Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK2387e191 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> SIP/2.0 180 Ringing From: "79166612794";tag=as790ff91d To: ;tag=c1cb190 Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 102 INVITE Server: vocl-essentra-bax/8.0.251 Contact: Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK2387e191 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/BAX-00000131 is ringing [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> SIP/2.0 180 Ringing From: "79166612794";tag=as790ff91d To: ;tag=c1cb190 Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 102 INVITE Server: vocl-essentra-bax/8.0.251 Contact: Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK2387e191 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/BAX-00000131 is ringing [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> SIP/2.0 200 OK From: "79166612794";tag=as790ff91d To: ;tag=c1cb190 Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 102 INVITE Server: vocl-essentra-bax/8.0.251 Contact: Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK2387e191 Content-Type: application/sdp Content-Length: 240 v=0 o=Essentra-Relay 1041641809 1041641809 IN IP4 82.204.187.10 s=- c=IN IP4 82.204.187.10 t=0 0 m=audio 41498 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (10 headers 11 lines) --- Found RTP audio format 18 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 82.204.187.10:41498 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 82.204.187.10, port 5060 Transmitting (no NAT) to 82.204.187.10:5060: ACK sip:74956986015@82.204.187.10;vtservice=CallControl.CallControlServlet SIP/2.0 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK541b1f52;rport Max-Forwards: 70 From: "79166612794" ;tag=as790ff91d To: ;tag=c1cb190 Contact: Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.17 Content-Length: 0 --- -- SIP/BAX-00000131 answered SIP/CX-00000130 -- Native bridging SIP/CX-00000130 and SIP/BAX-00000131 set_destination: Parsing for address/port to send to set_destination: set destination to 82.204.187.9, port 5060 Audio is at 192.168.0.28 port 19238 Adding codec 0x100 (g729) 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 82.204.187.9:5060: INVITE sip:+79166612794@82.204.187.9;user=phone;PG=MGTS SIP/2.0 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK1490042d;rport Max-Forwards: 70 From: ;tag=as4db5349f To: 79166612794 ;tag=80370848-0-13c4-45026-1e943-3a77dd38-1e943 Contact: Call-ID: 881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 332 v=0 o=root 236836765 236836766 IN IP4 82.204.187.10 s=Asterisk PBX 1.6.2.17 c=IN IP4 82.204.187.10 t=0 0 m=audio 41498 RTP/AVP 18 0 8 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- set_destination: Parsing for address/port to send to set_destination: set destination to 82.204.187.10, port 5060 Audio is at 192.168.0.28 port 18636 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 82.204.187.10:5060: INVITE sip:74956986015@82.204.187.10;vtservice=CallControl.CallControlServlet SIP/2.0 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK546e5dec;rport Max-Forwards: 70 From: "79166612794" ;tag=as790ff91d To: ;tag=c1cb190 Contact: Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 284 v=0 o=root 727601229 727601230 IN IP4 82.204.187.7 s=Asterisk PBX 1.6.2.17 c=IN IP4 82.204.187.7 t=0 0 m=audio 4220 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> SIP/2.0 100 Trying From: "79166612794";tag=as790ff91d To: ;tag=4f955d98 Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 103 INVITE Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK546e5dec Content-Length: 0 <-------------> --- (7 headers 0 lines) --- [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.9:5060 ---> SIP/2.0 200 OK From: ;tag=as4db5349f To: 79166612794 ;tag=80370848-0-13c4-45026-1e943-3a77dd38-1e943 Call-ID: 881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A CSeq: 102 INVITE P-Charging-Vector: icid-value=881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A; icid-generated-at=82.204.187.9 Server: vocl-essentra-cx/8.0F2-19020-48 Via: SIP/2.0/UDP 192.168.0.28:5060;rport=5060;branch=z9hG4bK1490042d Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=Essentra-CX-AC 3013260210 2 IN IP4 82.204.187.9 s=- c=IN IP4 82.204.187.7 t=0 0 m=audio 4220 RTP/AVP 18 96 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=silenceSupp:off - - - - a=sendrecv <-------------> --- (11 headers 12 lines) --- Found RTP audio format 18 Found RTP audio format 96 Found audio description format telephone-event for ID 96 Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 82.204.187.7:4220 set_destination: Parsing for address/port to send to set_destination: set destination to 82.204.187.9, port 5060 Transmitting (no NAT) to 82.204.187.9:5060: ACK sip:+79166612794@82.204.187.9;user=phone;PG=MGTS SIP/2.0 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK2956b6b5;rport Max-Forwards: 70 From: ;tag=as4db5349f To: 79166612794 ;tag=80370848-0-13c4-45026-1e943-3a77dd38-1e943 Contact: Call-ID: 881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.17 Content-Length: 0 --- [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> SIP/2.0 200 OK From: "79166612794";tag=as790ff91d To: ;tag=c1cb190 Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 103 INVITE Server: vocl-essentra-bax/8.0.251 Contact: Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK546e5dec Content-Type: application/sdp Content-Length: 240 v=0 o=Essentra-Relay 1041642249 1041642249 IN IP4 82.204.187.10 s=- c=IN IP4 82.204.187.10 t=0 0 m=audio 41498 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (10 headers 11 lines) --- Found RTP audio format 18 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 82.204.187.10:41498 set_destination: Parsing for address/port to send to set_destination: set destination to 82.204.187.10, port 5060 Transmitting (no NAT) to 82.204.187.10:5060: ACK sip:74956986015@82.204.187.10;vtservice=CallControl.CallControlServlet SIP/2.0 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK1148e519;rport Max-Forwards: 70 From: "79166612794" ;tag=as790ff91d To: ;tag=c1cb190 Contact: Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.2.17 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 82.204.187.10, port 5060 Audio is at 192.168.0.28 port 18636 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 82.204.187.10:5060: INVITE sip:74956986015@82.204.187.10;vtservice=CallControl.CallControlServlet SIP/2.0 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK418c7282;rport Max-Forwards: 70 From: "79166612794" ;tag=as790ff91d To: ;tag=c1cb190 Contact: Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 284 v=0 o=root 727601229 727601231 IN IP4 82.204.187.7 s=Asterisk PBX 1.6.2.17 c=IN IP4 82.204.187.7 t=0 0 m=audio 4220 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> SIP/2.0 100 Trying From: "79166612794";tag=as790ff91d To: ;tag=523ebb28 Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 104 INVITE Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK418c7282 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> SIP/2.0 200 OK From: "79166612794";tag=as790ff91d To: ;tag=c1cb190 Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 104 INVITE Server: vocl-essentra-bax/8.0.251 Contact: Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK418c7282 Content-Type: application/sdp Content-Length: 240 v=0 o=Essentra-Relay 1041642655 1041642655 IN IP4 82.204.187.10 s=- c=IN IP4 82.204.187.10 t=0 0 m=audio 41498 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (10 headers 11 lines) --- Found RTP audio format 18 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 82.204.187.10:41498 set_destination: Parsing for address/port to send to set_destination: set destination to 82.204.187.10, port 5060 Transmitting (no NAT) to 82.204.187.10:5060: ACK sip:74956986015@82.204.187.10;vtservice=CallControl.CallControlServlet SIP/2.0 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK7a1a39ea;rport Max-Forwards: 70 From: "79166612794" ;tag=as790ff91d To: ;tag=c1cb190 Contact: Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 104 ACK User-Agent: Asterisk PBX 1.6.2.17 Content-Length: 0 --- [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> INVITE sip:+79166612794@192.168.0.28 SIP/2.0 From: ;tag=c1cb190 To: "79166612794";tag=as790ff91d Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 120 INVITE Via: SIP/2.0/UDP 82.204.187.10:5060;rport;branch=z9hG4bK-530b0ae0-4d708dae;vtservice=CallControl.CallControlServlet Contact: Max-Forwards: 70 User-Agent: vocl-essentra-bax/8.0.251 Content-Type: application/sdp Content-Length: 213 v=0 o=Essentra-Relay 3508210734 3508210734 IN IP4 82.204.187.10 s=IVR MOH c=IN IP4 82.204.187.10 t=0 0 m=audio 41918 RTP/AVP 18 101 a=rtpmap:18 g729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (11 headers 9 lines) --- Sending to 82.204.187.10 : 5060 (no NAT) Found RTP audio format 18 Found RTP audio format 101 Found audio description format g729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 82.204.187.10:41918 <--- Transmitting (no NAT) to 82.204.187.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 82.204.187.10:5060;branch=z9hG4bK-530b0ae0-4d708dae;vtservice=CallControl.CallControlServlet;received=82.204.187.10;rport=5060 From: ;tag=c1cb190 To: "79166612794";tag=as790ff91d Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 120 INVITE Server: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 192.168.0.28 port 18636 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 82.204.187.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.204.187.10:5060;branch=z9hG4bK-530b0ae0-4d708dae;vtservice=CallControl.CallControlServlet;received=82.204.187.10;rport=5060 From: ;tag=c1cb190 To: "79166612794";tag=as790ff91d Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 120 INVITE Server: Asterisk PBX 1.6.2.17 [Kvoice_switch_0*CLI> [0KAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 284 v=0 o=root 727601229 727601232 IN IP4 82.204.187.7 s=Asterisk PBX 1.6.2.17 c=IN IP4 82.204.187.7 t=0 0 m=audio 4220 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> ACK sip:+79166612794@192.168.0.28 SIP/2.0 From: ;tag=c1cb190 To: "79166612794";tag=as790ff91d Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 120 ACK Via: SIP/2.0/UDP 82.204.187.10:5060;rport;branch=z9hG4bK-47b3fe-18171bdd-42377ea1 User-Agent: vocl-essentra-bax/8.0.251 Max-Forwards: 70 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> INVITE sip:+79166612794@192.168.0.28 SIP/2.0 From: ;tag=c1cb190 To: "79166612794";tag=as790ff91d Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 136 INVITE Via: SIP/2.0/UDP 82.204.187.10:5060;rport;branch=z9hG4bK-5554c628-4d708db9;vtservice=CallControl.CallControlServlet Contact: Max-Forwards: 70 User-Agent: vocl-essentra-bax/8.0.251 Content-Type: application/sdp Content-Length: 291 v=0 o=Essentra-Relay 386681775 386681661 IN IP4 82.204.187.10 s=Phone-Call c=IN IP4 82.204.187.10 t=0 0 m=audio 41878 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=rtcp:6001 IN IP4 192.168.2.5 <-------------> --- (11 headers 13 lines) --- Sending to 82.204.187.10 : 5060 (no NAT) Found RTP audio format 18 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 82.204.187.10:41878 <--- Transmitting (no NAT) to 82.204.187.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 82.204.187.10:5060;branch=z9hG4bK-5554c628-4d708db9;vtservice=CallControl.CallControlServlet;received=82.204.187.10;rport=5060 From: ;tag=c1cb190 To: "79166612794";tag=as790ff91d Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 136 INVITE Server: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 192.168.0.28 port 18636 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 82.204.187.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.204.187.10:5060;branch=z9hG4bK-5554c628-4d708db9;vtservice=CallControl.CallControlServlet;received=82.204.187.10;rport=5060 From: ;tag=c1cb190 To: "79166612794";tag=as790ff91d Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 136 INVITE Server: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 284 v=0 o=root 727601229 727601233 IN IP4 82.204.187.7 s=Asterisk PBX 1.6.2.17 c=IN IP4 82.204.187.7 t=0 0 m=audio 4220 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> ACK sip:+79166612794@192.168.0.28 SIP/2.0 From: ;tag=c1cb190 To: "79166612794";tag=as790ff91d Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 136 ACK Via: SIP/2.0/UDP 82.204.187.10:5060;rport;branch=z9hG4bK-47b40c-1817510d-7a66306b User-Agent: vocl-essentra-bax/8.0.251 Max-Forwards: 70 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.10:5060 ---> BYE sip:+79166612794@192.168.0.28 SIP/2.0 From: ;tag=c1cb190 To: "79166612794";tag=as790ff91d Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 152 BYE Via: SIP/2.0/UDP 82.204.187.10:5060;rport;branch=z9hG4bK-4e7abb18-4d708dc0;vtservice=CallControl.CallControlServlet Max-Forwards: 69 User-Agent: vocl-essentra-bax/8.0.251 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 82.204.187.10 : 5060 (no NAT) <--- Transmitting (no NAT) to 82.204.187.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.204.187.10:5060;branch=z9hG4bK-4e7abb18-4d708dc0;vtservice=CallControl.CallControlServlet;received=82.204.187.10;rport=5060 From: ;tag=c1cb190 To: "79166612794";tag=as790ff91d Call-ID: 39258aa51a58caf14924c7c94493ac30@192.168.0.28 CSeq: 152 BYE Server: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 82.204.187.9, port 5060 Audio is at 192.168.0.28 port 19238 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 82.204.187.9:5060: INVITE sip:+79166612794@82.204.187.9;user=phone;PG=MGTS SIP/2.0 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK4edd380c;rport Max-Forwards: 70 From: ;tag=as4db5349f To: 79166612794 ;tag=80370848-0-13c4-45026-1e943-3a77dd38-1e943 Contact: Call-ID: 881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 282 v=0 o=root 236836765 236836767 IN IP4 192.168.0.28 s=Asterisk PBX 1.6.2.17 c=IN IP4 192.168.0.28 t=0 0 m=audio 19238 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 4 09:59:14] [0;31mERROR[0m[29013]: [1;37mcdr_sqlite3_custom.c[0m:[1;37m269[0m [1;37mwrite_cdr[0m: unable to open database file. SQL: INSERT INTO cdr (calldate,clid,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,test) VALUES ('2011-03-04 09:58:47','"79166612794" <+79166612794>','fromCX','SIP/CX-00000130','SIP/BAX-00000131','Dial','SIP/BAX/74956986015','27','23','ANSWERED','DOCUMENTATION','','1299221927.304','',''). [Kvoice_switch_0*CLI> [0K == Spawn extension (fromCX, +74956986015, 3) exited non-zero on 'SIP/CX-00000130' Scheduling destruction of SIP dialog '881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A' in 32000 ms (Method: ACK) [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.9:5060 ---> SIP/2.0 200 OK From: ;tag=as4db5349f To: 79166612794 ;tag=80370848-0-13c4-45026-1e943-3a77dd38-1e943 Call-ID: 881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A CSeq: 103 INVITE P-Charging-Vector: icid-value=881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A; icid-generated-at=82.204.187.9 Server: vocl-essentra-cx/8.0F2-19020-48 Via: SIP/2.0/UDP 192.168.0.28:5060;rport=5060;branch=z9hG4bK4edd380c Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=Essentra-CX-AC 3013260210 3 IN IP4 82.204.187.9 s=- c=IN IP4 82.204.187.7 t=0 0 m=audio 4220 RTP/AVP 18 96 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=silenceSupp:off - - - - a=sendrecv <-------------> --- (11 headers 12 lines) --- Found RTP audio format 18 Found RTP audio format 96 Found audio description format telephone-event for ID 96 Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 82.204.187.7:4220 set_destination: Parsing for address/port to send to set_destination: set destination to 82.204.187.9, port 5060 Transmitting (no NAT) to 82.204.187.9:5060: ACK sip:+79166612794@82.204.187.9;user=phone;PG=MGTS SIP/2.0 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK4eab771c;rport Max-Forwards: 70 From: ;tag=as4db5349f To: 79166612794 ;tag=80370848-0-13c4-45026-1e943-3a77dd38-1e943 Contact: Call-ID: 881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.2.17 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 82.204.187.9, port 5060 Reliably Transmitting (no NAT) to 82.204.187.9:5060: BYE sip:+79166612794@82.204.187.9;user=phone;PG=MGTS SIP/2.0 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK675a673a;rport Max-Forwards: 70 From: ;tag=as4db5349f To: 79166612794 ;tag=80370848-0-13c4-45026-1e943-3a77dd38-1e943 Call-ID: 881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A CSeq: 104 BYE User-Agent: Asterisk PBX 1.6.2.17 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A' in 32000 ms (Method: ACK) Really destroying SIP dialog '39258aa51a58caf14924c7c94493ac30@192.168.0.28' Method: BYE [Kvoice_switch_0*CLI> [0K <--- SIP read from UDP:82.204.187.9:5060 ---> SIP/2.0 200 OK From: ;tag=as4db5349f To: 79166612794 ;tag=80370848-0-13c4-45026-1e943-3a77dd38-1e943 Call-ID: 881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A CSeq: 104 BYE Via: SIP/2.0/UDP 192.168.0.28:5060;rport=5060;branch=z9hG4bK675a673a Server: vocl-essentra-cx/8.0F2-19020-48 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '881B9C9D-0178-42A7-8AA2-29EA6F9D6F3A' Method: ACK [Kvoice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> voice_switch_0*CLI> exit [0m]0;root@voice_switch_0:~[root@voice_switch_0 ~]# ]0;root@voice_switch_0:~[root@voice_switch_0 ~]# ]0;root@voice_switch_0:~[root@voice_switch_0 ~]# ]0;root@voice_switch_0:~[root@voice_switch_0 ~]# ]0;root@voice_switch_0:~[root@voice_switch_0 ~]# ]0;root@voice_switch_0:~[root@voice_switch_0 ~]# ]0;root@voice_switch_0:~[root@voice_switch_0 ~]# ]0;root@voice_switch_0:~[root@voice_switch_0 ~]# ]0;root@voice_switch_0:~[root@voice_switch_0 ~]# ]0;root@voice_switch_0:~[root@voice_switch_0 ~]#