Script started on Wed 09 Jan 2008 02:46:37 PM BRST PBGW6:/usr/src/asterisk-1.4.4# asterisk -r Asterisk 1.4.4, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.4.4 currently running on PBGW6 (pid = 5724) PBGW6*CLI> Verbosity is at least 3 PBGW6*CLI> <--- SIP read from 172.20.0.38:5080 ---> INVITE sip:00111111004884195169008441311@172.25.1.226:5060 SIP/2.0 From: "106";tag=e836a04d To: Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 1 INVITE Content-Length: 445 Content-Type: application/sdp Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318416 Contact: Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: eyeBeam release 1013t stamp 43069 Referred-By: Call-Info: ; MzNmNmYxNDQ4ZjJjODRjY2M5MjQ2M2NhMzdiMTVhMmI.;gen-rt=e836a04d;gen-lt=A016DB95-E951-47C9-B275-7026530959BB-164406 Session-Expires: 1800;refresher=uac Min-SE: 90 Supported: timer v=0 o=- 0 2 IN IP4 172.25.9.253 s=CounterPath eyeBeam 1.5 c=IN IP4 172.25.9.253 t=0 0 m=audio 65294 RTP/AVP 0 8 18 3 101 a=alt:1 3 : 49e0JDKJ Ebaug1Mt 169.254.192.189 65294 a=alt:2 2 : TULWu3xh 2nAuqEqc 169.254.252.106 65294 a=alt:3 1 : 21yX0ezh wcqG15DD 172.25.9.253 65294 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:BC706AB3449E4DB58E1A87B387915345 <-------------> --- (17 headers 15 lines) --- Sending to 172.20.0.38 : 5080 (no NAT) Using INVITE request as basis request - 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 Found peer 'SipServer_Genesys_DIB' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 172.25.9.253:65294 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.25.9.253:65294 Looking for 00111111004884195169008441311 in ramal-genesys (domain 172.25.1.226) list_route: hop: <--- Transmitting (no NAT) to 172.20.0.38:5080 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318416;received=172.20.0.38 From: "106";tag=e836a04d To: Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [00111111004884195169008441311@ramal-genesys:1] Set("SIP/Ramal_Genesys-08450c00", "cd_telefone=008441311") in new stack -- Executing [00111111004884195169008441311@ramal-genesys:2] Set("SIP/Ramal_Genesys-08450c00", "cd_operacao=00111") in new stack -- Executing [00111111004884195169008441311@ramal-genesys:3] Set("SIP/Ramal_Genesys-08450c00", "telefone=84195169") in new stack -- Executing [00111111004884195169008441311@ramal-genesys:4] Set("SIP/Ramal_Genesys-08450c00", "ddd=48") in new stack -- Executing [00111111004884195169008441311@ramal-genesys:5] Set("SIP/Ramal_Genesys-08450c00", "tipo_origem=3") in new stack -- Executing [00111111004884195169008441311@ramal-genesys:6] Set("SIP/Ramal_Genesys-08450c00", "localidade=") in new stack -- Executing [00111111004884195169008441311@ramal-genesys:7] Set("SIP/Ramal_Genesys-08450c00", "localidade=0") in new stack -- Executing [00111111004884195169008441311@ramal-genesys:8] Set("SIP/Ramal_Genesys-08450c00", "localidade=0") in new stack -- Executing [00111111004884195169008441311@ramal-genesys:9] Set("SIP/Ramal_Genesys-08450c00", "localidade=1") in new stack -- Executing [00111111004884195169008441311@ramal-genesys:10] Set("SIP/Ramal_Genesys-08450c00", "fac=00") in new stack -- Executing [00111111004884195169008441311@ramal-genesys:11] Set("SIP/Ramal_Genesys-08450c00", "dial_string=00111111004884195169008441311") in new stack -- Executing [00111111004884195169008441311@ramal-genesys:12] Set("SIP/Ramal_Genesys-08450c00", "chave_gravacao=1199897230.0") in new stack PBGW6*CLI> -- Executing [00111111004884195169008441311@ramal-genesys:13] Set("SIP/Ramal_Genesys-08450c00", "dstgravacao=2008-01-09") in new stack PBGW6*CLI> -- Executing [00111111004884195169008441311@ramal-genesys:14] Set("SIP/Ramal_Genesys-08450c00", "path=/var/spool/asterisk/monitor") in new stack PBGW6*CLI> -- Executing [00111111004884195169008441311@ramal-genesys:15] Set("SIP/Ramal_Genesys-08450c00", "status=LIGACAO") in new stack PBGW6*CLI> -- Executing [00111111004884195169008441311@ramal-genesys:16] Set("SIP/Ramal_Genesys-08450c00", "tp_lgc=local") in new stack PBGW6*CLI> -- Executing [00111111004884195169008441311@ramal-genesys:17] Set("SIP/Ramal_Genesys-08450c00", "procedure=ODBC_SQL_SANTANDER_PREDITIVO") in new stack PBGW6*CLI> -- Executing [00111111004884195169008441311@ramal-genesys:18] GotoIf("SIP/Ramal_Genesys-08450c00", "0?desvia") in new stack PBGW6*CLI> -- Executing [00111111004884195169008441311@ramal-genesys:19] GotoIf("SIP/Ramal_Genesys-08450c00", "0?desvia") in new stack PBGW6*CLI> -- Executing [00111111004884195169008441311@ramal-genesys:20] GotoIf("SIP/Ramal_Genesys-08450c00", "0?hangup") in new stack PBGW6*CLI> -- Executing [00111111004884195169008441311@ramal-genesys:21] Set("SIP/Ramal_Genesys-08450c00", "dial_string=00111111214884195169008441311") in new stack PBGW6*CLI> -- Executing [00111111004884195169008441311@ramal-genesys:22] Macro("SIP/Ramal_Genesys-08450c00", "pred_grava_ligacao|48|84195169|008441311|00111|ODBC_SQL_SANTANDER_PREDITIVO|1199897230.0|2008-01-09|/var/spool/asterisk/monitor|LIGACAO") in new stack PBGW6*CLI> -- Executing [s@macro-pred_grava_ligacao:1] System("SIP/Ramal_Genesys-08450c00", "sh /etc/asterisk/check_dir.sh") in new stack PBGW6*CLI> -- Executing [s@macro-pred_grava_ligacao:2] Set("SIP/Ramal_Genesys-08450c00", "dstgravacao=2008-01-09") in new stack -- Executing [s@macro-pred_grava_ligacao:3] Set("SIP/Ramal_Genesys-08450c00", "ddd=48") in new stack -- Executing [s@macro-pred_grava_ligacao:4] Set("SIP/Ramal_Genesys-08450c00", "telefone=84195169") in new stack -- Executing [s@macro-pred_grava_ligacao:5] Set("SIP/Ramal_Genesys-08450c00", "cd_telefone=008441311") in new stack -- Executing [s@macro-pred_grava_ligacao:6] Set("SIP/Ramal_Genesys-08450c00", "cd_operacao=00111") in new stack -- Executing [s@macro-pred_grava_ligacao:7] Set("SIP/Ramal_Genesys-08450c00", "procedure=ODBC_SQL_SANTANDER_PREDITIVO") in new stack -- Executing [s@macro-pred_grava_ligacao:8] Set("SIP/Ramal_Genesys-08450c00", "chave_gravacao=1199897230.0") in new stack -- Executing [s@macro-pred_grava_ligacao:9] Set("SIP/Ramal_Genesys-08450c00", "dstgravacao=2008-01-09") in new stack -- Executing [s@macro-pred_grava_ligacao:10] Set("SIP/Ramal_Genesys-08450c00", "path=/var/spool/asterisk/monitor") in new stack -- Executing [s@macro-pred_grava_ligacao:11] Set("SIP/Ramal_Genesys-08450c00", "status=LIGACAO") in new stack -- Executing [s@macro-pred_grava_ligacao:12] NoOp("SIP/Ramal_Genesys-08450c00", "gravacao=ODBC_SQL_SANTANDER_PREDITIVO(48|84195169|008441311|LIGACAO") in new stack PBGW6*CLI> -- Executing [s@macro-pred_grava_ligacao:13] Set("SIP/Ramal_Genesys-08450c00", "gravacao=000000007098671") in new stack PBGW6*CLI> -- Executing [s@macro-pred_grava_ligacao:14] Set("SIP/Ramal_Genesys-08450c00", "nm_grava=00111_000000007098671") in new stack PBGW6*CLI> -- Executing [s@macro-pred_grava_ligacao:15] GotoIf("SIP/Ramal_Genesys-08450c00", "0?desvia:continua") in new stack PBGW6*CLI> -- Goto (macro-pred_grava_ligacao,s,19) PBGW6*CLI> -- Executing [s@macro-pred_grava_ligacao:19] MixMonitor("SIP/Ramal_Genesys-08450c00", "2008-01-09/temp/00111_000000007098671.WAV|W(1)") in new stack PBGW6*CLI> -- Executing [s@macro-pred_grava_ligacao:20] System("SIP/Ramal_Genesys-08450c00", "echo "#!/bin/bash" >> /var/spool/asterisk/monitor/2008-01-09/temp/1199897230.0.sh") in new stack PBGW6*CLI> == Begin MixMonitor Recording SIP/Ramal_Genesys-08450c00 PBGW6*CLI> -- Executing [s@macro-pred_grava_ligacao:21] System("SIP/Ramal_Genesys-08450c00", "echo mv /var/spool/asterisk/monitor/2008-01-09/temp/00111_000000007098671.WAV /var/spool/asterisk/monitor/2008-01-09/00111_000000007098671.WAV >> /var/spool/asterisk/monitor/2008-01-09/temp/1199897230.0.sh") in new stack PBGW6*CLI> -- Executing [s@macro-pred_grava_ligacao:22] Ringing("SIP/Ramal_Genesys-08450c00", "") in new stack PBGW6*CLI> <--- Transmitting (no NAT) to 172.20.0.38:5080 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318416;received=172.20.0.38 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces C PBGW6*CLI> ontact: Content-Length: 0 <------------> PBGW6*CLI> -- Executing [s@macro-pred_grava_ligacao:23] NoOp("SIP/Ramal_Genesys-08450c00", "Vai para originador com gravacao ativada") in new stack PBGW6*CLI> -- Executing [00111111004884195169008441311@ramal-genesys:23] Goto("SIP/Ramal_Genesys-08450c00", "local") in new stack PBGW6*CLI> -- Goto (ramal-genesys,00111111004884195169008441311,26) PBGW6*CLI> -- Executing [00111111004884195169008441311@ramal-genesys:26] Dial("SIP/Ramal_Genesys-08450c00", "zap/g21/84195169|30|tT") in new stack PBGW6*CLI> -- Requested transfer capability: 0x00 - SPEECH PBGW6*CLI> -- Called g21/84195169 PBGW6*CLI> Audio is at 172.25.1.226 port 10046 PBGW6*CLI> Adding codec 0x8 (alaw) to SDP PBGW6*CLI> Adding non-codec 0x1 (telephone-event) to SDP PBGW6*CLI> <--- Transmitting (no NAT) to 172.20.0.38:5080 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318416;received=172.20.0.38 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 5724 5724 IN IP4 172.25.1.226 s=session c=IN IP4 172.25.1.226 t=0 0 m=audio 10046 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 <------------> PBGW6*CLI> -- Zap/1-1 is proceeding passing it to SIP/Ramal_Genesys-08450c00 PBGW6*CLI> -- Zap/1-1 is ringing <--- Transmitting (no NAT) to 172.20.0.38:5080 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318416;received=172.20.0.38 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> PBGW6*CLI> -- Zap/1-1 answered SIP/Ramal_Genesys-08450c00 Audio is at 172.25.1.226 port 10046 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 172.20.0.38:5080 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318416;received=172.20.0.38 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 5724 5725 IN IP4 172.25.1.226 s=session c=IN IP4 172.25.1.226 t=0 0 m=audio 10046 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 PBGW6*CLI> =sendrecv <------------> PBGW6*CLI> <--- SIP read from 172.20.0.38:5080 ---> ACK sip:00111111004884195169008441311@172.25.1.226 SIP/2.0 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 1 ACK Content-Length: 0 Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318425 Max-Forwards: 69 User-Agent: eyeBeam release 1013t stamp 43069 <-------------> --- (9 headers 0 lines) --- PBGW6*CLI> <--- SIP read from 172.20.0.38:5080 ---> INVITE sip:00111111004884195169008441311@172.25.1.226 SIP/2.0 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 2 INVITE Content-Length: 444 Content-Type: application/sdp Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318430 Contact: Max-Forwards: 70 Session-Expires: 1800;refresher=uac Min-SE: 90 Supported: 100rel,timer v=0 o=GSIP-TREATMENT 1199897531 1 IN IP4 172.20.0.38 s=Genesys SIP Server c=IN IP4 0.0.0.0 t=0 0 m=audio 65294 RTP/AVP 0 8 18 3 101 a=alt:1 3 : 49e0JDKJ Ebaug1Mt 169.254.192.189 65294 a=alt:2 2 : TULWu3xh 2nAuqEqc 169.254.252.106 65294 a=alt:3 1 : 21yX0ezh wcqG15DD 172.25.9.253 65294 a=x-rtp-session-id:BC706AB3449E4DB58E1A87B387915345 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (13 headers 14 lines) --- Sending to 172.20.0.38 : 5080 (no NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:65294 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 0.0.0.0:65294 Audio is at 172.25.1.226 port 10046 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 172.20.0.38:5080 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318430;received=172.20.0.38 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 5724 5726 IN IP4 172.25.1.226 s=session c=IN IP4 172.25.1.226 t=0 0 m=audio 10046 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 <------------> PBGW6*CLI> -- Started music on hold, class 'default', on Zap/1-1 PBGW6*CLI> <--- SIP read from 172.20.0.38:5080 ---> ACK sip:00111111004884195169008441311@172.25.1.226 SIP/2.0 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 2 ACK Content-Length: 0 Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318431 <-------------> --- (7 headers 0 lines) --- PBGW6*CLI> <--- SIP read from 172.20.0.38:5080 ---> INVITE sip:00111111004884195169008441311@172.25.1.226 SIP/2.0 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 3 INVITE Content-Length: 232 Content-Type: application/sdp Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318435 Contact: Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: eyeBeam release 1013t stamp 43069 Referred-By: Call-Info: ; MzNmNmYxNDQ4ZjJjODRjY2M5MjQ2M2NhMzdiMTVhMmI.;gen-rt=e836a04d;gen-lt=A016DB95-E951-47C9-B275-7026530959BB-164406 Session-Expires: 1800;refresher=uac Min-SE: 90 Supported: timer v=0 o=- 0 4 IN IP4 172.25.9.253 s=CounterPath eyeBeam 1.5 c=IN IP4 0.0.0.0 t=0 0 m=audio 65294 RTP/AVP 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendonly a=x-rtp-session-id:BC706AB3449E4DB58E1A87B387915345 <-------------> --- (17 headers 10 lines) --- Sending to 172.20.0.38 : 5080 (no NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:65294 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 0.0.0.0:65294 Audio is at 172.25.1.226 port 10046 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 172.20.0.38:5080 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318435;received=172.20.0.38 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 5724 5727 IN IP4 172.25.1.226 s=session c=IN IP4 172.25.1.226 t=0 0 m=audio 10046 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=recvonly <------------> PBGW6*CLI> -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 PBGW6*CLI> <--- SIP read from 172.20.0.38:5080 ---> ACK sip:00111111004884195169008441311@172.25.1.226 SIP/2.0 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 3 ACK Content-Length: 0 Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318438 Max-Forwards: 69 User-Agent: eyeBeam release 1013t stamp 43069 <-------------> --- (9 headers 0 lines) --- PBGW6*CLI> <--- SIP read from 172.20.0.38:5080 ---> INVITE sip:00111111004884195169008441311@172.25.1.226 SIP/2.0 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 4 INVITE Content-Length: 0 Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318443 Contact: Max-Forwards: 70 Session-Expires: 1800;refresher=uac Min-SE: 90 Supported: 100rel,timer <-------------> --- (12 headers 0 lines) --- Sending to 172.20.0.38 : 5080 (no NAT) Audio is at 172.25.1.226 port 10046 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 172.20.0.38:5080 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318443;received=172.20.0.38 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 4 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 5724 5728 IN IP4 172.25.1.226 s=session c=IN IP4 172.25.1.226 t=0 0 m=audio 10046 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=recvonly <------------> PBGW6*CLI> <--- SIP read from 172.20.0.38:5080 ---> ACK sip:00111111004884195169008441311@172.25.1.226 SIP/2.0 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 4 ACK Content-Length: 240 Content-Type: application/sdp Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318446 Call-Info: ; 879C703A-E5E2-48D9-B07B-E00D819F10B1-189871%40172.20.0.38;gen-rt=as28ba4d11;gen-lt=7c2b5312 v=0 o=root 14465 14467 IN IP4 172.25.1.225 s=session c=IN IP4 172.25.1.225 t=0 0 m=audio 17746 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 <-------------> --- (9 headers 12 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 172.25.1.225:17746 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.25.1.225:17746 PBGW6*CLI> -- Stopped music on hold on Zap/1-1 PBGW6*CLI> <--- SIP read from 172.20.0.38:5080 ---> BYE sip:00111111004884195169008441311@172.25.1.226 SIP/2.0 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 5 BYE Content-Length: 0 Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318450 User-Agent: Asterisk PBX Max-Forwards: 69 <-------------> --- (9 headers 0 lines) --- PBGW6*CLI> Sending to 172.20.0.38 : 5080 (no NAT) <--- Transmitting (no NAT) to 172.20.0.38:5080 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.20.0.38:5080;branch=z9hG4bK95DC915C-3228-4DDD-B016-A3457A843ACA-318450;received=172.20.0.38 From: "106";tag=e836a04d To: ;tag=as4f419f65 Call-ID: 879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38 CSeq: 5 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> PBGW6*CLI> -- Hungup 'Zap/1-1' == Spawn extension (ramal-genesys, 00111111004884195169008441311, 26) exited non-zero on 'SIP/Ramal_Genesys-08450c00' -- Executing [h@ramal-genesys:1] GotoIf("SIP/Ramal_Genesys-08450c00", "0?hang") in new stack -- Executing [h@ramal-genesys:2] GotoIf("SIP/Ramal_Genesys-08450c00", "0?hang") in new stack -- Executing [h@ramal-genesys:3] TrySystem("SIP/Ramal_Genesys-08450c00", "sh /var/spool/asterisk/monitor/2008-01-09/temp/1199897230.0.sh") in new stack PBGW6*CLI> -- Executing [h@ramal-genesys:4] TrySystem("SIP/Ramal_Genesys-08450c00", "rm /var/spool/asterisk/monitor/2008-01-09/temp/1199897230.0.sh") in new stack PBGW6*CLI> -- Executing [h@ramal-genesys:5] Hangup("SIP/Ramal_Genesys-08450c00", "") in new stack PBGW6*CLI> == Spawn extension (ramal-genesys, h, 5) exited non-zero on 'SIP/Ramal_Genesys-08450c00' PBGW6*CLI> == End MixMonitor Recording SIP/Ramal_Genesys-08450c00 PBGW6*CLI> Really destroying SIP dialog '879C703A-E5E2-48D9-B07B-E00D819F10B1-189865@172.20.0.38' Method: BYE PBGW6*CLI> exit PBGW6:/usr/src/asterisk-1.4.4# Script done on Wed 09 Jan 2008 02:47:46 PM BRST