Sip read: INVITE sip:K0000000336926317@212.130.0.200 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 217.116.243.234;branch=z9hG4bKf8b7.64b768c5.0 Via: SIP/2.0/UDP gw1.v2tel.net:5060;rport=5060;received=212.130.0.195;branch=z9hG4bK-vega1-000A-0001-09F5-ABB138B1 From: 026153060 ;tag=0006-08D4-36D9DD03 To: Max-Forwards: 69 Call-ID: 0004-0180-8F75B73F-0@gw1.v2tel.net CSeq: 51954758 INVITE Contact: Supported: replaces, privacy Allow: INVITE,ACK,BYE,CANCEL,INFO,NOTIFY,OPTIONS,REFER Accept-Language: en Content-Type: application/sdp User-Agent: Vega100-T1E1/08.02.05.1xT019 Content-Length: 303 v=0 o=Vega50 4921 1 IN IP4 212.130.0.195 s=Sip Call t=0 0 m=audio 35278 RTP/AVP 18 4 0 8 96 c=IN IP4 217.116.243.234 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15,16 a=direction:active a=nortpproxy:yes 16 headers, 14 lines Using latest request as basis request Sending to 217.116.243.234 : 5060 (non-NAT) Found peer 'sipout-b-36926317' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 96 Peer audio RTP is at port 217.116.243.234:35278 Found description format G729 Found description format G723 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for K0000000336926317 in sipin list_route: hop: list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.116.243.234;branch=z9hG4bKf8b7.64b768c5.0 Via: SIP/2.0/UDP gw1.v2tel.net:5060;received=212.130.0.195;branch=z9hG4bK-vega1-000A-0001-09F5-ABB138B1 From: 026153060 ;tag=0006-08D4-36D9DD03 To: ;tag=as2e44f162 Call-ID: 0004-0180-8F75B73F-0@gw1.v2tel.net CSeq: 51954758 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 217.116.243.234:5060 -- Executing ODBCget("SIP/36926317-d1c7", "days=WEEKDAY/05072005") in new stack -- odbcget: varname=days, family=WEEKDAY, key=05072005 > app_dbodbc: Query Successful! -- odbcget: set variable days to tue -- Executing Goto("SIP/36926317-d1c7", "3") in new stack -- Goto (sipin,K0000000336926317,3) -- Executing ODBCget("SIP/36926317-d1c7", "hours=K00000003/OPENING36926317tue") in new stack -- odbcget: varname=hours, family=K00000003, key=OPENING36926317tue > app_dbodbc: Query Successful! -- odbcget: Value not found in database. -- Executing SetVar("SIP/36926317-d1c7", "hours=*") in new stack -- Executing Goto("SIP/36926317-d1c7", "5") in new stack -- Goto (sipin,K0000000336926317,5) -- Executing ODBCget("SIP/36926317-d1c7", "overrule=K00000003/NIGHTOVERRULE") in new stack -- odbcget: varname=overrule, family=K00000003, key=NIGHTOVERRULE > app_dbodbc: Query Successful! -- odbcget: Value not found in database. -- Executing SetVar("SIP/36926317-d1c7", "nightoverrule=no") in new stack -- Executing Goto("SIP/36926317-d1c7", "7") in new stack -- Goto (sipin,K0000000336926317,7) -- Executing GotoIfTime("SIP/36926317-d1c7", "*|*|*|*|?10") in new stack -- Goto (sipin,K0000000336926317,10) -- Executing SetVar("SIP/36926317-d1c7", "time=DAY") in new stack -- Executing SetVar("SIP/36926317-d1c7", "number=local36926317") in new stack -- Executing GotoIf("SIP/36926317-d1c7", "0?13:14") in new stack -- Goto (sipin,K0000000336926317,14) -- Executing SetCIDNum("SIP/36926317-d1c7", "0026153060") in new stack -- Executing SetCallerID("SIP/36926317-d1c7", "0026153060") in new stack -- Executing ODBCget("SIP/36926317-d1c7", "number=K00000003/INCOMINGDAY36926317") in new stack -- odbcget: varname=number, family=K00000003, key=INCOMINGDAY36926317 > app_dbodbc: Query Successful! -- odbcget: Value not found in database. -- Executing ODBCget("SIP/36926317-d1c7", "number=K00000003/INCOMINGDAY3692631700") in new stack -- odbcget: varname=number, family=K00000003, key=INCOMINGDAY3692631700 > app_dbodbc: Query Successful! -- odbcget: Value not found in database. -- Executing ODBCget("SIP/36926317-d1c7", "number=K00000003/INCOMINGDAY36926317002") in new stack -- odbcget: varname=number, family=K00000003, key=INCOMINGDAY36926317002 > app_dbodbc: Query Successful! -- odbcget: Value not found in database. -- Executing ODBCget("SIP/36926317-d1c7", "number=K00000003/INCOMINGDAY369263170026") in new stack -- odbcget: varname=number, family=K00000003, key=INCOMINGDAY369263170026 > app_dbodbc: Query Successful! -- odbcget: Value not found in database. -- Executing ODBCget("SIP/36926317-d1c7", "number=K00000003/INCOMINGDAY3692631700261") in new stack -- odbcget: varname=number, family=K00000003, key=INCOMINGDAY3692631700261 > app_dbodbc: Query Successful! -- odbcget: Value not found in database. -- Executing Goto("SIP/36926317-d1c7", "K00000003|local36926317|1") in new stack -- Goto (K00000003,local36926317,1) -- Executing Macro("SIP/36926317-d1c7", "exttypeA|317|SIP/100000003317|K00000003|0") in new stack -- Executing SetVar("SIP/36926317-d1c7", "exttimeout=20") in new stack -- Executing ODBCget("SIP/36926317-d1c7", "exttimeout=K00000003/MYPHONE317EXTTIMEOUT") in new stack -- odbcget: varname=exttimeout, family=K00000003, key=MYPHONE317EXTTIMEOUT > app_dbodbc: Query Successful! -- odbcget: Value not found in database. -- Executing Goto("SIP/36926317-d1c7", "s|4") in new stack -- Goto (macro-exttypeA,s,4) -- Executing GotoIf("SIP/36926317-d1c7", "0?5:7") in new stack -- Goto (macro-exttypeA,s,7) -- Executing SetVar("SIP/36926317-d1c7", "mystatus=") in new stack -- Executing GotoIf("SIP/36926317-d1c7", "0?500:9") in new stack -- Goto (macro-exttypeA,s,9) -- Executing ODBCget("SIP/36926317-d1c7", "knownnumber1=K00000003/MYPHONE317MOBILE") in new stack -- odbcget: varname=knownnumber1, family=K00000003, key=MYPHONE317MOBILE > app_dbodbc: Query Successful! -- odbcget: Value not found in database. -- Executing ODBCget("SIP/36926317-d1c7", "knownnumber2=K00000003/MYPHONE317HOME") in new stack -- odbcget: varname=knownnumber2, family=K00000003, key=MYPHONE317HOME > app_dbodbc: Query Successful! -- odbcget: Value not found in database. -- Executing SetVar("SIP/36926317-d1c7", "incoming=0026153060") in new stack Jul 5 16:34:45 WARNING[9850]: ast_expr.y:475 ast_yyerror: ast_yyerror(): syntax error: syntax error; Input: = 0026153060 ^^^^^ ^ -- Executing GotoIf("SIP/36926317-d1c7", "0?700:13") in new stack -- Goto (macro-exttypeA,s,13) Jul 5 16:34:45 WARNING[9850]: ast_expr.y:475 ast_yyerror: ast_yyerror(): syntax error: syntax error; Input: = 0026153060 ^^^^^ ^ -- Executing GotoIf("SIP/36926317-d1c7", "0?700:14") in new stack -- Goto (macro-exttypeA,s,14) -- Executing ODBCget("SIP/36926317-d1c7", "temp=K00000003/MYPHONE317CFIM") in new stack -- odbcget: varname=temp, family=K00000003, key=MYPHONE317CFIM > app_dbodbc: Query Successful! -- odbcget: Value not found in database. -- Executing Goto("SIP/36926317-d1c7", "s|18") in new stack -- Goto (macro-exttypeA,s,18) -- Executing ODBCget("SIP/36926317-d1c7", "temp=K00000003/MYPHONE317JOIN") in new stack -- odbcget: varname=temp, family=K00000003, key=MYPHONE317JOIN > app_dbodbc: Query Successful! -- odbcget: Value not found in database. -- Executing Goto("SIP/36926317-d1c7", "s|23") in new stack -- Goto (macro-exttypeA,s,23) -- Executing Dial("SIP/36926317-d1c7", "SIP/100000003317|20|ft") in new stack We're at 212.130.0.200 port 10406 Answering/Requesting with root capability 0x8 (alaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 13 lines Reliably Transmitting: INVITE sip:100000003317@130.228.38.64;user=phone SIP/2.0 Via: SIP/2.0/UDP 212.130.0.200:5060;branch=z9hG4bK61f28dc6;rport From: "0026153060" ;tag=as27211501 To: Contact: Call-ID: 612f56756b63d97f35f635ce027459c0@212.130.0.200 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 05 Jul 2005 14:34:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 289 v=0 o=root 9850 9850 IN IP4 212.130.0.200 s=session c=IN IP4 212.130.0.200 t=0 0 m=audio 10406 RTP/AVP 8 18 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 130.228.38.64:5060 -- Called 100000003317 blade1*CLI> Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.130.0.200:5060;branch=z9hG4bK61f28dc6;rport From: "0026153060" ;tag=as27211501 To: ;tag=1c1078031736 Call-ID: 612f56756b63d97f35f635ce027459c0@212.130.0.200 CSeq: 102 INVITE Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.40.181.314 Content-Length: 0 10 headers, 0 lines blade1*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 212.130.0.200:5060;branch=z9hG4bK61f28dc6;rport From: "0026153060" ;tag=as27211501 To: ;tag=1c1078031736 Call-ID: 612f56756b63d97f35f635ce027459c0@212.130.0.200 CSeq: 102 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.40.181.314 Content-Length: 0 11 headers, 0 lines -- SIP/100000003317-dc81 is ringing Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 217.116.243.234;branch=z9hG4bKf8b7.64b768c5.0 Via: SIP/2.0/UDP gw1.v2tel.net:5060;received=212.130.0.195;branch=z9hG4bK-vega1-000A-0001-09F5-ABB138B1 From: 026153060 ;tag=0006-08D4-36D9DD03 To: ;tag=as2e44f162 Call-ID: 0004-0180-8F75B73F-0@gw1.v2tel.net CSeq: 51954758 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 217.116.243.234:5060 blade1*CLI> Sip read: REGISTER sip:datelco.v2tel.net SIP/2.0 Via: SIP/2.0/UDP 80.163.71.242:23081;branch=z9hG4bK-mj1h8tsbj7az;rport From: "Service" ;tag=9cvq33aix5 To: "Service" Call-ID: 3c2670059ec1-qh1kawbdszvn@192-168-0-152 CSeq: 6037 REGISTER Max-Forwards: 70 Contact: ;q=1.0 User-Agent: snom200-3.56m P-NAT-Refresh: 15 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.0.152 WWW-Contact: WWW-Contact: Expires: 60 Content-Length: 0 17 headers, 0 lines Using latest request as basis request Sending to 80.163.71.242 : 23081 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 80.163.71.242:23081;branch=z9hG4bK-mj1h8tsbj7az;received=80.163.71.242;rport=23081 From: "Service" ;tag=9cvq33aix5 To: "Service" ;tag=as2f3734a6 Call-ID: 3c2670059ec1-qh1kawbdszvn@192-168-0-152 CSeq: 6037 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 80.163.71.242:23081 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 80.163.71.242:23081;branch=z9hG4bK-mj1h8tsbj7az;received=80.163.71.242;rport=23081 From: "Service" ;tag=9cvq33aix5 To: "Service" ;tag=as2f3734a6 Call-ID: 3c2670059ec1-qh1kawbdszvn@192-168-0-152 CSeq: 6037 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: WWW-Authenticate: Digest realm="asterisk", nonce="7b48bd5e" Content-Length: 0 to 80.163.71.242:23081 Scheduling destruction of call '3c2670059ec1-qh1kawbdszvn@192-168-0-152' in 15000 ms blade1*CLI> Sip read: REGISTER sip:datelco.v2tel.net SIP/2.0 Via: SIP/2.0/UDP 80.163.71.242:23081;branch=z9hG4bK-ppuzq8glhhlk;rport From: "Service" ;tag=9cvq33aix5 To: "Service" Call-ID: 3c2670059ec1-qh1kawbdszvn@192-168-0-152 CSeq: 6038 REGISTER Max-Forwards: 70 Contact: ;q=1.0 User-Agent: snom200-3.56m P-NAT-Refresh: 15 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.0.152 WWW-Contact: WWW-Contact: Authorization: Digest username="100000020782",realm="asterisk",nonce="7b48bd5e",uri="sip:datelco.v2tel.net",response="7ef570f00f7f3565ff777142644b96ce",algorithm=md5 Expires: 60 Content-Length: 0 18 headers, 0 lines Using latest request as basis request Sending to 80.163.71.242 : 23081 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 80.163.71.242:23081;branch=z9hG4bK-ppuzq8glhhlk;received=80.163.71.242;rport=23081 From: "Service" ;tag=9cvq33aix5 To: "Service" ;tag=as2f3734a6 Call-ID: 3c2670059ec1-qh1kawbdszvn@192-168-0-152 CSeq: 6038 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 80.163.71.242:23081 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 80.163.71.242:23081;branch=z9hG4bK-ppuzq8glhhlk;received=80.163.71.242;rport=23081 From: "Service" ;tag=9cvq33aix5 To: "Service" ;tag=as2f3734a6 Call-ID: 3c2670059ec1-qh1kawbdszvn@192-168-0-152 CSeq: 6038 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 60 Contact: ;expires=60 Date: Tue, 05 Jul 2005 14:34:46 GMT Content-Length: 0 to 80.163.71.242:23081 Scheduling destruction of call '3c2670059ec1-qh1kawbdszvn@192-168-0-152' in 15000 ms Destroying call '3c36857a858d-332i2o20sg5b@192-168-100-111' blade1*CLI> Sip read: REGISTER sip:datelco.v2tel.net SIP/2.0 Via: SIP/2.0/UDP 80.163.71.242:23064;branch=z9hG4bK-f0i9ac03o707;rport From: "Omstillingen" ;tag=p25wxyggdu To: "Omstillingen" Call-ID: 3c267005afa7-j4gynxpb8qrh@192-168-0-150 CSeq: 6127 REGISTER Max-Forwards: 70 Contact: ;q=1.0 User-Agent: snom200-3.56m P-NAT-Refresh: 15 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.0.150 WWW-Contact: WWW-Contact: Expires: 60 Content-Length: 0 17 headers, 0 lines Using latest request as basis request Sending to 80.163.71.242 : 23064 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 80.163.71.242:23064;branch=z9hG4bK-f0i9ac03o707;received=80.163.71.242;rport=23356 From: "Omstillingen" ;tag=p25wxyggdu To: "Omstillingen" ;tag=as184324a5 Call-ID: 3c267005afa7-j4gynxpb8qrh@192-168-0-150 CSeq: 6127 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 80.163.71.242:23356 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 80.163.71.242:23064;branch=z9hG4bK-f0i9ac03o707;received=80.163.71.242;rport=23356 From: "Omstillingen" ;tag=p25wxyggdu To: "Omstillingen" ;tag=as184324a5 Call-ID: 3c267005afa7-j4gynxpb8qrh@192-168-0-150 CSeq: 6127 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: WWW-Authenticate: Digest realm="asterisk", nonce="0009a8fa" Content-Length: 0 to 80.163.71.242:23356 Scheduling destruction of call '3c267005afa7-j4gynxpb8qrh@192-168-0-150' in 15000 ms blade1*CLI> Sip read: REGISTER sip:datelco.v2tel.net SIP/2.0 Via: SIP/2.0/UDP 80.163.71.242:23064;branch=z9hG4bK-tg5cv9lvb31s;rport From: "Omstillingen" ;tag=p25wxyggdu To: "Omstillingen" Call-ID: 3c267005afa7-j4gynxpb8qrh@192-168-0-150 CSeq: 6128 REGISTER Max-Forwards: 70 Contact: ;q=1.0 User-Agent: snom200-3.56m P-NAT-Refresh: 15 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.0.150 WWW-Contact: WWW-Contact: Authorization: Digest username="100000020780",realm="asterisk",nonce="0009a8fa",uri="sip:datelco.v2tel.net",response="f82953ddaa74b396c24131400c0cdd65",algorithm=md5 Expires: 60 Content-Length: 0 18 headers, 0 lines Using latest request as basis request Sending to 80.163.71.242 : 23064 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 80.163.71.242:23064;branch=z9hG4bK-tg5cv9lvb31s;received=80.163.71.242;rport=23356 From: "Omstillingen" ;tag=p25wxyggdu To: "Omstillingen" ;tag=as184324a5 Call-ID: 3c267005afa7-j4gynxpb8qrh@192-168-0-150 CSeq: 6128 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 80.163.71.242:23356 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 80.163.71.242:23064;branch=z9hG4bK-tg5cv9lvb31s;received=80.163.71.242;rport=23356 From: "Omstillingen" ;tag=p25wxyggdu To: "Omstillingen" ;tag=as184324a5 Call-ID: 3c267005afa7-j4gynxpb8qrh@192-168-0-150 CSeq: 6128 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 60 Contact: ;expires=60 Date: Tue, 05 Jul 2005 14:34:50 GMT Content-Length: 0 to 80.163.71.242:23356 Scheduling destruction of call '3c267005afa7-j4gynxpb8qrh@192-168-0-150' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:100000003317@130.228.38.64;user=phone SIP/2.0 Via: SIP/2.0/UDP 212.130.0.200:5060;branch=z9hG4bK0091fb19;rport From: "asterisk" ;tag=as64dd2ded To: Contact: Call-ID: 3a13e5924e6a456f78b0dabf5479673b@212.130.0.200 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 42 Messages-Waiting: no Voice-Message: 0/0 (NAT) to 130.228.38.64:5060 Scheduling destruction of call '3a13e5924e6a456f78b0dabf5479673b@212.130.0.200' in 15000 ms blade1*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.130.0.200:5060;branch=z9hG4bK0091fb19;rport From: "asterisk" ;tag=as64dd2ded To: ;tag=1c186057702 Call-ID: 3a13e5924e6a456f78b0dabf5479673b@212.130.0.200 CSeq: 102 NOTIFY Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.40.181.314 Content-Length: 0 11 headers, 0 lines Destroying call '3a13e5924e6a456f78b0dabf5479673b@212.130.0.200' blade1*CLI> Sip read: REGISTER sip:datelco.v2tel.net SIP/2.0 Via: SIP/2.0/UDP 80.163.71.242:23079;branch=z9hG4bK-49g36asiagae;rport From: "Salg" ;tag=9ewwtpgiii To: "Salg" Call-ID: 3c267005a275-cx5d6vaggsfv@192-168-0-151 CSeq: 6031 REGISTER Max-Forwards: 70 Contact: ;q=1.0 User-Agent: snom200-3.56m P-NAT-Refresh: 15 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.0.151 WWW-Contact: WWW-Contact: Expires: 60 Content-Length: 0 17 headers, 0 lines Using latest request as basis request Sending to 80.163.71.242 : 23079 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 80.163.71.242:23079;branch=z9hG4bK-49g36asiagae;received=80.163.71.242;rport=23079 From: "Salg" ;tag=9ewwtpgiii To: "Salg" ;tag=as2d8690bf Call-ID: 3c267005a275-cx5d6vaggsfv@192-168-0-151 CSeq: 6031 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 80.163.71.242:23079 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 80.163.71.242:23079;branch=z9hG4bK-49g36asiagae;received=80.163.71.242;rport=23079 From: "Salg" ;tag=9ewwtpgiii To: "Salg" ;tag=as2d8690bf Call-ID: 3c267005a275-cx5d6vaggsfv@192-168-0-151 CSeq: 6031 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: WWW-Authenticate: Digest realm="asterisk", nonce="2d191463" Content-Length: 0 to 80.163.71.242:23079 Scheduling destruction of call '3c267005a275-cx5d6vaggsfv@192-168-0-151' in 15000 ms blade1*CLI> Sip read: REGISTER sip:datelco.v2tel.net SIP/2.0 Via: SIP/2.0/UDP 80.163.71.242:23079;branch=z9hG4bK-dnnuvyuwr8h8;rport From: "Salg" ;tag=9ewwtpgiii To: "Salg" Call-ID: 3c267005a275-cx5d6vaggsfv@192-168-0-151 CSeq: 6032 REGISTER Max-Forwards: 70 Contact: ;q=1.0 User-Agent: snom200-3.56m P-NAT-Refresh: 15 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.0.151 WWW-Contact: WWW-Contact: Authorization: Digest username="100000020781",realm="asterisk",nonce="2d191463",uri="sip:datelco.v2tel.net",response="a105926fde562e189fdcbc47adfa9878",algorithm=md5 Expires: 60 Content-Length: 0 18 headers, 0 lines Using latest request as basis request Sending to 80.163.71.242 : 23079 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 80.163.71.242:23079;branch=z9hG4bK-dnnuvyuwr8h8;received=80.163.71.242;rport=23079 From: "Salg" ;tag=9ewwtpgiii To: "Salg" ;tag=as2d8690bf Call-ID: 3c267005a275-cx5d6vaggsfv@192-168-0-151 CSeq: 6032 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 80.163.71.242:23079 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 80.163.71.242:23079;branch=z9hG4bK-dnnuvyuwr8h8;received=80.163.71.242;rport=23079 From: "Salg" ;tag=9ewwtpgiii To: "Salg" ;tag=as2d8690bf Call-ID: 3c267005a275-cx5d6vaggsfv@192-168-0-151 CSeq: 6032 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 60 Contact: ;expires=60 Date: Tue, 05 Jul 2005 14:34:52 GMT Content-Length: 0 to 80.163.71.242:23079 Scheduling destruction of call '3c267005a275-cx5d6vaggsfv@192-168-0-151' in 15000 ms blade1*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.130.0.200:5060;branch=z9hG4bK61f28dc6;rport From: "0026153060" ;tag=as27211501 To: ;tag=1c1078031736 Call-ID: 612f56756b63d97f35f635ce027459c0@212.130.0.200 CSeq: 102 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.40.181.314 Content-Type: application/sdp Content-Length: 224 v=0 o=AudiocodesGW 839731 475398 IN IP4 130.228.38.64 s=Phone-Call c=IN IP4 130.228.38.64 t=0 0 m=audio 4000 RTP/AVP 8 96 a=rtpmap:8 pcma/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv 12 headers, 11 lines Found RTP audio format 8 Found RTP audio format 96 Peer audio RTP is at port 130.228.38.64:4000 Found description format pcma Found description format telephone-event Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 130.228.38.64, port 5060 Transmitting: ACK sip:100000003317@130.228.38.64;user=phone SIP/2.0 Via: SIP/2.0/UDP 212.130.0.200:5060;branch=z9hG4bK72ae5778;rport From: "0026153060" ;tag=as27211501 To: ;tag=1c1078031736 Contact: Call-ID: 612f56756b63d97f35f635ce027459c0@212.130.0.200 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 130.228.38.64:5060 -- SIP/100000003317-dc81 answered SIP/36926317-d1c7 We're at 212.130.0.200 port 10586 Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 217.116.243.234;branch=z9hG4bKf8b7.64b768c5.0 Via: SIP/2.0/UDP gw1.v2tel.net:5060;received=212.130.0.195;branch=z9hG4bK-vega1-000A-0001-09F5-ABB138B1 Record-Route: From: 026153060 ;tag=0006-08D4-36D9DD03 To: ;tag=as2e44f162 Call-ID: 0004-0180-8F75B73F-0@gw1.v2tel.net CSeq: 51954758 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 286 v=0 o=root 9850 9850 IN IP4 212.130.0.200 s=session c=IN IP4 212.130.0.200 t=0 0 m=audio 10586 RTP/AVP 8 18 3 0 96 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - to 217.116.243.234:5060 -- Attempting native bridge of SIP/36926317-d1c7 and SIP/100000003317-dc81 blade1*CLI> Sip read: ACK sip:K0000000336926317@212.130.0.200 SIP/2.0 Via: SIP/2.0/UDP 217.116.243.234;branch=0 Via: SIP/2.0/UDP gw1.v2tel.net:5060;rport=5060;received=212.130.0.195 From: 026153060 ;tag=0006-08D4-36D9DD03 To: ;tag=as2e44f162 Max-Forwards: 69 Call-ID: 0004-0180-8F75B73F-0@gw1.v2tel.net CSeq: 51954758 ACK Contact: User-Agent: Vega100-T1E1/08.02.05.1xT019 Content-Length: 0 11 headers, 0 lines 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:100000020782@80.163.71.242:23081;line=zhewd0cz SIP/2.0 Via: SIP/2.0/UDP 212.130.0.200:5060;branch=z9hG4bK7e84bfd4;rport From: "asterisk" ;tag=as30ad985e To: Contact: Call-ID: 3c54c9e446c6563f562ddbd43dffc8f4@212.130.0.200 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 42 Messages-Waiting: no Voice-Message: 0/0 (NAT) to 80.163.71.242:23081 Scheduling destruction of call '3c54c9e446c6563f562ddbd43dffc8f4@212.130.0.200' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:100000020781@80.163.71.242:23079;line=k0jjy13r SIP/2.0 Via: SIP/2.0/UDP 212.130.0.200:5060;branch=z9hG4bK59c8f523;rport From: "asterisk" ;tag=as2448a964 To: Contact: Call-ID: 5e4ee96b32d5f631491a00c242639da9@212.130.0.200 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 42 Messages-Waiting: no Voice-Message: 0/0 (NAT) to 80.163.71.242:23079 Scheduling destruction of call '5e4ee96b32d5f631491a00c242639da9@212.130.0.200' in 15000 ms blade1*CLI> Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 212.130.0.200:5060;branch=z9hG4bK7e84bfd4;rport=5060 From: "asterisk" ;tag=as30ad985e To: Call-ID: 3c54c9e446c6563f562ddbd43dffc8f4@212.130.0.200 CSeq: 102 NOTIFY Content-Length: 0 7 headers, 0 lines Destroying call '3c54c9e446c6563f562ddbd43dffc8f4@212.130.0.200' blade1*CLI> Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 212.130.0.200:5060;branch=z9hG4bK59c8f523;rport=5060 From: "asterisk" ;tag=as2448a964 To: Call-ID: 5e4ee96b32d5f631491a00c242639da9@212.130.0.200 CSeq: 102 NOTIFY Content-Length: 0 7 headers, 0 lines Destroying call '5e4ee96b32d5f631491a00c242639da9@212.130.0.200' blade1*CLI> Sip read: BYE sip:0026153060@212.130.0.200 SIP/2.0 Via: SIP/2.0/UDP 130.228.38.64;branch=z9hG4bKacglQTyBf Max-Forwards: 70 From: ;tag=1c1078031736 To: "0026153060" ;tag=as27211501 Call-ID: 612f56756b63d97f35f635ce027459c0@212.130.0.200 CSeq: 1 BYE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.40.181.314 Content-Length: 0 12 headers, 0 lines Sending to 130.228.38.64 : 5060 (NAT) Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 130.228.38.64;branch=z9hG4bKacglQTyBf;received=130.228.38.64;rport=5060 From: ;tag=1c1078031736 To: "0026153060" ;tag=as27211501 Call-ID: 612f56756b63d97f35f635ce027459c0@212.130.0.200 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 130.228.38.64:5060 == Spawn extension (macro-exttypeA, s, 23) exited non-zero on 'SIP/36926317-d1c7' in macro 'exttypeA' == Spawn extension (K00000003, local36926317, 1) exited non-zero on 'SIP/36926317-d1c7' set_destination: Parsing for address/port to send to set_destination: set destination to 217.116.243.234, port 5060 Reliably Transmitting: BYE sip:026153060@212.130.0.195:5060 SIP/2.0 Via: SIP/2.0/UDP 212.130.0.200:5060;branch=z9hG4bK3b8fa959 Route: From: ;tag=as2e44f162 To: 026153060 ;tag=0006-08D4-36D9DD03 Contact: Call-ID: 0004-0180-8F75B73F-0@gw1.v2tel.net CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 217.116.243.234:5060 blade1*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.130.0.200:5060;branch=z9hG4bK3b8fa959 Record-route: From: ;tag=as2e44f162 To: 026153060 ;tag=0006-08D4-36D9DD03 Call-ID: 0004-0180-8F75B73F-0@gw1.v2tel.net CSeq: 102 BYE Contact: User-Agent: Vega100-T1E1/08.02.05.1xT019 Content-Length: 0 10 headers, 0 lines Message is BYE Destroying call '612f56756b63d97f35f635ce027459c0@212.130.0.200' Destroying call '0004-0180-8F75B73F-0@gw1.v2tel.net' Destroying call '2509716861YZVz@130.228.38.64' 11 headers, 2 lines