[Sep 10 10:42:59] VERBOSE[1834] logger.c: == Parsing '/etc/asterisk/manager.conf': [Sep 10 10:42:59] VERBOSE[1834] logger.c: Found [Sep 10 10:42:59] VERBOSE[1834] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': [Sep 10 10:42:59] VERBOSE[1834] logger.c: Found [Sep 10 10:42:59] VERBOSE[1834] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': [Sep 10 10:42:59] VERBOSE[1834] logger.c: Found [Sep 10 10:42:59] VERBOSE[1834] logger.c: == Manager 'inftester' logged on from 10.215.147.112 [Sep 10 10:42:59] VERBOSE[1834] logger.c: == Manager 'inftester' logged off from 10.215.147.112 [Sep 10 10:42:59] VERBOSE[1417] logger.c: -- Remote UNIX connection [Sep 10 10:43:02] VERBOSE[1436] logger.c: Reliably Transmitting (NAT) to 10.215.146.175:5060: OPTIONS sip:4064@10.215.146.175:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.215.147.112:5060;branch=z9hG4bK0c15015f;rport From: "Unknown" ;tag=as3885ce8d To: Contact: Call-ID: 496845870714586154df05760cff5d89@10.215.147.112 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 10 Sep 2009 08:43:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- [Sep 10 10:43:02] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.175:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.215.147.112:5060;branch=z9hG4bK0c15015f;rport From: "Unknown" ;tag=as3885ce8d To: ;tag=42c642a8819468ed Call-ID: 496845870714586154df05760cff5d89@10.215.147.112 CSeq: 102 OPTIONS User-Agent: Grandstream GXP280 1.2.1.4 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Sep 10 10:43:02] VERBOSE[1436] logger.c: --- (11 headers 0 lines) --- [Sep 10 10:43:02] VERBOSE[1436] logger.c: Really destroying SIP dialog '496845870714586154df05760cff5d89@10.215.147.112' Method: OPTIONS [Sep 10 10:43:02] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> INVITE sip:4@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK957ef0b7a9c454bd From: "TEST2" ;tag=e9445d071df0bf70 To: Contact: Supported: replaces, timer, path Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2671 INVITE User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 408 v=0 o=4063 8000 8000 IN IP4 10.215.146.165 s=SIP Call c=IN IP4 10.215.146.165 t=0 0 m=audio 5028 RTP/AVP 0 8 4 18 2 97 9 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Sep 10 10:43:02] VERBOSE[1436] logger.c: --- (13 headers 19 lines) --- [Sep 10 10:43:02] VERBOSE[1436] logger.c: Sending to 10.215.146.165 : 5060 (no NAT) [Sep 10 10:43:02] VERBOSE[1436] logger.c: Using INVITE request as basis request - 2fa63c881788d281@10.215.146.165 [Sep 10 10:43:02] VERBOSE[1436] logger.c: <--- Reliably Transmitting (NAT) to 10.215.146.165:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK957ef0b7a9c454bd;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2671 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="76cb03f1" Content-Length: 0 <------------> [Sep 10 10:43:02] VERBOSE[1436] logger.c: Scheduling destruction of SIP dialog '2fa63c881788d281@10.215.146.165' in 32000 ms (Method: INVITE) [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found user '4063' [Sep 10 10:43:02] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> ACK sip:4@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK957ef0b7a9c454bd From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: path Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2671 ACK User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:02] VERBOSE[1436] logger.c: --- (12 headers 0 lines) --- [Sep 10 10:43:02] WARNING[1436] chan_sip.c: PEDANTIC TESTING processing ACK... [Sep 10 10:43:02] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> INVITE sip:4@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK8e92439580e7b7e7 From: "TEST2" ;tag=e9445d071df0bf70 To: Contact: Supported: replaces, timer, path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:4@10.215.147.112", nonce="76cb03f1", response="26b739dd7be3787799dc082f1fb8161a" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2672 INVITE User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 408 v=0 o=4063 8000 8001 IN IP4 10.215.146.165 s=SIP Call c=IN IP4 10.215.146.165 t=0 0 m=audio 5028 RTP/AVP 0 8 4 18 2 97 9 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Sep 10 10:43:02] VERBOSE[1436] logger.c: --- (14 headers 19 lines) --- [Sep 10 10:43:02] VERBOSE[1436] logger.c: Sending to 10.215.146.165 : 5060 (NAT) [Sep 10 10:43:02] VERBOSE[1436] logger.c: Using INVITE request as basis request - 2fa63c881788d281@10.215.146.165 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found user '4063' [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found RTP audio format 0 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found RTP audio format 8 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found RTP audio format 4 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found RTP audio format 18 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found RTP audio format 2 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found RTP audio format 97 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found RTP audio format 9 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found RTP audio format 3 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found RTP audio format 101 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Peer audio RTP is at port 10.215.146.165:5028 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found audio description format PCMU for ID 0 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found audio description format PCMA for ID 8 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found audio description format G723 for ID 4 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found audio description format G729 for ID 18 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found audio description format G726-32 for ID 2 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found audio description format iLBC for ID 97 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found audio description format G722 for ID 9 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found audio description format GSM for ID 3 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Found audio description format telephone-event for ID 101 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Capabilities: us - 0x2 (gsm), peer - audio=0x1d0f (g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing), combined - 0x2 (gsm) [Sep 10 10:43:02] VERBOSE[1436] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Sep 10 10:43:02] VERBOSE[1436] logger.c: Peer audio RTP is at port 10.215.146.165:5028 [Sep 10 10:43:02] DEBUG[1436] chan_sip.c: Call from peer '4063' is 1 out of 50 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Looking for 4 in from-internal (domain 10.215.147.112) [Sep 10 10:43:02] VERBOSE[1436] logger.c: <--- Reliably Transmitting (NAT) to 10.215.146.165:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK8e92439580e7b7e7;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2672 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Sep 10 10:43:02] DEBUG[1436] chan_sip.c: Call from peer '4063' removed from call limit 50 [Sep 10 10:43:02] VERBOSE[1436] logger.c: Scheduling destruction of SIP dialog '2fa63c881788d281@10.215.146.165' in 32000 ms (Method: INVITE) [Sep 10 10:43:02] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> ACK sip:4@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK8e92439580e7b7e7 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:4@10.215.147.112", nonce="76cb03f1", response="26b739dd7be3787799dc082f1fb8161a" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2672 ACK User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:02] VERBOSE[1436] logger.c: --- (13 headers 0 lines) --- [Sep 10 10:43:02] WARNING[1436] chan_sip.c: PEDANTIC TESTING processing ACK... [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> INVITE sip:40@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKffadc3579d4d2061 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: replaces, timer, path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:40@10.215.147.112", nonce="76cb03f1", response="be427d34da161c8ce8c773ddde15851b" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2673 INVITE User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 408 v=0 o=4063 8000 8002 IN IP4 10.215.146.165 s=SIP Call c=IN IP4 10.215.146.165 t=0 0 m=audio 5030 RTP/AVP 0 8 4 18 2 97 9 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Sep 10 10:43:03] VERBOSE[1436] logger.c: --- (14 headers 19 lines) --- [Sep 10 10:43:03] VERBOSE[1436] logger.c: Sending to 10.215.146.165 : 5060 (NAT) [Sep 10 10:43:03] VERBOSE[1436] logger.c: Using INVITE request as basis request - 2fa63c881788d281@10.215.146.165 [Sep 10 10:43:03] NOTICE[1436] chan_sip.c: Correct auth, but based on stale nonce received from ';tag=as1211b496' [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- Reliably Transmitting (NAT) to 10.215.146.165:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKffadc3579d4d2061;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2673 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6cb0d599", stale=true Content-Length: 0 <------------> [Sep 10 10:43:03] VERBOSE[1436] logger.c: Scheduling destruction of SIP dialog '2fa63c881788d281@10.215.146.165' in 32000 ms (Method: INVITE) [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found user '4063' [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> ACK sip:40@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKffadc3579d4d2061 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496;tag=as1211b496 Contact: Supported: path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:40@10.215.147.112", nonce="76cb03f1", response="be427d34da161c8ce8c773ddde15851b" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2673 ACK User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:03] VERBOSE[1436] logger.c: --- (13 headers 0 lines) --- [Sep 10 10:43:03] WARNING[1436] chan_sip.c: PEDANTIC TESTING processing ACK... [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> INVITE sip:40@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK4dd06fc9fdc9b2e0 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: replaces, timer, path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:40@10.215.147.112", nonce="6cb0d599", response="81b24ecf564716ec5b017a5a64ec6e65" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2674 INVITE User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 408 v=0 o=4063 8000 8003 IN IP4 10.215.146.165 s=SIP Call c=IN IP4 10.215.146.165 t=0 0 m=audio 5030 RTP/AVP 0 8 4 18 2 97 9 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Sep 10 10:43:03] VERBOSE[1436] logger.c: --- (14 headers 19 lines) --- [Sep 10 10:43:03] VERBOSE[1436] logger.c: Sending to 10.215.146.165 : 5060 (NAT) [Sep 10 10:43:03] VERBOSE[1436] logger.c: Using INVITE request as basis request - 2fa63c881788d281@10.215.146.165 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found user '4063' [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 0 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 8 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 4 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 18 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 2 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 97 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 9 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 3 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 101 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Peer audio RTP is at port 10.215.146.165:5030 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format PCMU for ID 0 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format PCMA for ID 8 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format G723 for ID 4 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format G729 for ID 18 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format G726-32 for ID 2 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format iLBC for ID 97 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format G722 for ID 9 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format GSM for ID 3 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format telephone-event for ID 101 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Capabilities: us - 0x2 (gsm), peer - audio=0x1d0f (g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing), combined - 0x2 (gsm) [Sep 10 10:43:03] VERBOSE[1436] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Sep 10 10:43:03] VERBOSE[1436] logger.c: Peer audio RTP is at port 10.215.146.165:5030 [Sep 10 10:43:03] DEBUG[1436] chan_sip.c: Call from peer '4063' is 1 out of 50 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Looking for 40 in from-internal (domain 10.215.147.112) [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- Reliably Transmitting (NAT) to 10.215.146.165:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK4dd06fc9fdc9b2e0;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2674 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Sep 10 10:43:03] DEBUG[1436] chan_sip.c: Call from peer '4063' removed from call limit 50 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Scheduling destruction of SIP dialog '2fa63c881788d281@10.215.146.165' in 32000 ms (Method: INVITE) [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> ACK sip:40@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK4dd06fc9fdc9b2e0 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:40@10.215.147.112", nonce="6cb0d599", response="81b24ecf564716ec5b017a5a64ec6e65" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2674 ACK User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:03] VERBOSE[1436] logger.c: --- (13 headers 0 lines) --- [Sep 10 10:43:03] WARNING[1436] chan_sip.c: PEDANTIC TESTING processing ACK... [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> INVITE sip:406@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK7f527b4ede770694 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: replaces, timer, path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:406@10.215.147.112", nonce="6cb0d599", response="bebda3b440ff08588e5dd461b2d32df9" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2675 INVITE User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 408 v=0 o=4063 8000 8004 IN IP4 10.215.146.165 s=SIP Call c=IN IP4 10.215.146.165 t=0 0 m=audio 5032 RTP/AVP 0 8 4 18 2 97 9 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Sep 10 10:43:03] VERBOSE[1436] logger.c: --- (14 headers 19 lines) --- [Sep 10 10:43:03] VERBOSE[1436] logger.c: Sending to 10.215.146.165 : 5060 (NAT) [Sep 10 10:43:03] VERBOSE[1436] logger.c: Using INVITE request as basis request - 2fa63c881788d281@10.215.146.165 [Sep 10 10:43:03] NOTICE[1436] chan_sip.c: Correct auth, but based on stale nonce received from ';tag=as1211b496' [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- Reliably Transmitting (NAT) to 10.215.146.165:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK7f527b4ede770694;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2675 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c6489bf", stale=true Content-Length: 0 <------------> [Sep 10 10:43:03] VERBOSE[1436] logger.c: Scheduling destruction of SIP dialog '2fa63c881788d281@10.215.146.165' in 32000 ms (Method: INVITE) [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found user '4063' [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> ACK sip:406@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK7f527b4ede770694 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496;tag=as1211b496 Contact: Supported: path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:406@10.215.147.112", nonce="6cb0d599", response="bebda3b440ff08588e5dd461b2d32df9" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2675 ACK User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:03] VERBOSE[1436] logger.c: --- (13 headers 0 lines) --- [Sep 10 10:43:03] WARNING[1436] chan_sip.c: PEDANTIC TESTING processing ACK... [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> INVITE sip:406@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKf51e12c449875e81 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: replaces, timer, path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:406@10.215.147.112", nonce="3c6489bf", response="34190bca99b67f41576cdd47bb4b0f47" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2676 INVITE User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 408 v=0 o=4063 8000 8005 IN IP4 10.215.146.165 s=SIP Call c=IN IP4 10.215.146.165 t=0 0 m=audio 5032 RTP/AVP 0 8 4 18 2 97 9 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Sep 10 10:43:03] VERBOSE[1436] logger.c: --- (14 headers 19 lines) --- [Sep 10 10:43:03] VERBOSE[1436] logger.c: Sending to 10.215.146.165 : 5060 (NAT) [Sep 10 10:43:03] VERBOSE[1436] logger.c: Using INVITE request as basis request - 2fa63c881788d281@10.215.146.165 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found user '4063' [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 0 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 8 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 4 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 18 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 2 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 97 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 9 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 3 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 101 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Peer audio RTP is at port 10.215.146.165:5032 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format PCMU for ID 0 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format PCMA for ID 8 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format G723 for ID 4 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format G729 for ID 18 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format G726-32 for ID 2 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format iLBC for ID 97 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format G722 for ID 9 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format GSM for ID 3 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format telephone-event for ID 101 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Capabilities: us - 0x2 (gsm), peer - audio=0x1d0f (g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing), combined - 0x2 (gsm) [Sep 10 10:43:03] VERBOSE[1436] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Sep 10 10:43:03] VERBOSE[1436] logger.c: Peer audio RTP is at port 10.215.146.165:5032 [Sep 10 10:43:03] DEBUG[1436] chan_sip.c: Call from peer '4063' is 1 out of 50 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Looking for 406 in from-internal (domain 10.215.147.112) [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- Reliably Transmitting (NAT) to 10.215.146.165:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKf51e12c449875e81;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2676 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Sep 10 10:43:03] DEBUG[1436] chan_sip.c: Call from peer '4063' removed from call limit 50 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Scheduling destruction of SIP dialog '2fa63c881788d281@10.215.146.165' in 32000 ms (Method: INVITE) [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> ACK sip:406@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKf51e12c449875e81 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:406@10.215.147.112", nonce="3c6489bf", response="34190bca99b67f41576cdd47bb4b0f47" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2676 ACK User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:03] VERBOSE[1436] logger.c: --- (13 headers 0 lines) --- [Sep 10 10:43:03] WARNING[1436] chan_sip.c: PEDANTIC TESTING processing ACK... [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> INVITE sip:4064@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK308f80966077ab67 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: replaces, timer, path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:4064@10.215.147.112", nonce="3c6489bf", response="df1c7738c82d9326c3f4125254935d31" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2677 INVITE User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 408 v=0 o=4063 8000 8006 IN IP4 10.215.146.165 s=SIP Call c=IN IP4 10.215.146.165 t=0 0 m=audio 5034 RTP/AVP 0 8 4 18 2 97 9 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Sep 10 10:43:03] VERBOSE[1436] logger.c: --- (14 headers 19 lines) --- [Sep 10 10:43:03] VERBOSE[1436] logger.c: Sending to 10.215.146.165 : 5060 (NAT) [Sep 10 10:43:03] VERBOSE[1436] logger.c: Using INVITE request as basis request - 2fa63c881788d281@10.215.146.165 [Sep 10 10:43:03] NOTICE[1436] chan_sip.c: Correct auth, but based on stale nonce received from ';tag=as1211b496' [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- Reliably Transmitting (NAT) to 10.215.146.165:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK308f80966077ab67;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2677 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09d47485", stale=true Content-Length: 0 <------------> [Sep 10 10:43:03] VERBOSE[1436] logger.c: Scheduling destruction of SIP dialog '2fa63c881788d281@10.215.146.165' in 32000 ms (Method: INVITE) [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found user '4063' [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> ACK sip:4064@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK308f80966077ab67 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496;tag=as1211b496 Contact: Supported: path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:4064@10.215.147.112", nonce="3c6489bf", response="df1c7738c82d9326c3f4125254935d31" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2677 ACK User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:03] VERBOSE[1436] logger.c: --- (13 headers 0 lines) --- [Sep 10 10:43:03] WARNING[1436] chan_sip.c: PEDANTIC TESTING processing ACK... [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> INVITE sip:4064@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKe8b028b0a963e236 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: replaces, timer, path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:4064@10.215.147.112", nonce="09d47485", response="ca824a855fe5adf9e652de99e1ae2f2e" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 INVITE User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 408 v=0 o=4063 8000 8007 IN IP4 10.215.146.165 s=SIP Call c=IN IP4 10.215.146.165 t=0 0 m=audio 5034 RTP/AVP 0 8 4 18 2 97 9 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Sep 10 10:43:03] VERBOSE[1436] logger.c: --- (14 headers 19 lines) --- [Sep 10 10:43:03] VERBOSE[1436] logger.c: Sending to 10.215.146.165 : 5060 (NAT) [Sep 10 10:43:03] VERBOSE[1436] logger.c: Using INVITE request as basis request - 2fa63c881788d281@10.215.146.165 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found user '4063' [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 0 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 8 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 4 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 18 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 2 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 97 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 9 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 3 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found RTP audio format 101 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Peer audio RTP is at port 10.215.146.165:5034 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format PCMU for ID 0 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format PCMA for ID 8 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format G723 for ID 4 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format G729 for ID 18 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format G726-32 for ID 2 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format iLBC for ID 97 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format G722 for ID 9 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format GSM for ID 3 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Found audio description format telephone-event for ID 101 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Capabilities: us - 0x2 (gsm), peer - audio=0x1d0f (g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing), combined - 0x2 (gsm) [Sep 10 10:43:03] VERBOSE[1436] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Sep 10 10:43:03] VERBOSE[1436] logger.c: Peer audio RTP is at port 10.215.146.165:5034 [Sep 10 10:43:03] DEBUG[1436] chan_sip.c: Call from peer '4063' is 1 out of 50 [Sep 10 10:43:03] VERBOSE[1436] logger.c: Looking for 4064 in from-internal (domain 10.215.147.112) [Sep 10 10:43:03] VERBOSE[1436] logger.c: list_route: hop: [Sep 10 10:43:03] VERBOSE[1436] logger.c: <--- Transmitting (NAT) to 10.215.146.165:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKe8b028b0a963e236;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [4064@from-internal:1] Macro("SIP/4063-08298a58", "exten-vm|novm|4064") in new stack [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-exten-vm:1] Macro("SIP/4063-08298a58", "user-callerid") in new stack [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-user-callerid:1] Set("SIP/4063-08298a58", "AMPUSER=4063") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: Set [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/4063-08298a58", "0?report") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: GotoIf [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/4063-08298a58", "1|Set|REALCALLERIDNUM=4063") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: ExecIf [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-user-callerid:4] Set("SIP/4063-08298a58", "AMPUSER=4063") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: Set [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-user-callerid:5] Set("SIP/4063-08298a58", "AMPUSERCIDNAME=INF TEST2") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: Set [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/4063-08298a58", "0?report") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: GotoIf [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-user-callerid:7] Set("SIP/4063-08298a58", "AMPUSERCID=4063") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: Set [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-user-callerid:8] Set("SIP/4063-08298a58", "CALLERID(all)="INF TEST2" <4063>") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: Set [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-user-callerid:9] Set("SIP/4063-08298a58", "REALCALLERIDNUM=4063") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: Set [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-user-callerid:10] ExecIf("SIP/4063-08298a58", "1|Set|CHANNEL(language)=es") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: ExecIf [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Last app: Set|CHANNEL(language)=es [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-user-callerid:11] GotoIf("SIP/4063-08298a58", "0?continue") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: GotoIf [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-user-callerid:12] Set("SIP/4063-08298a58", "__TTL=64") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: Set [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-user-callerid:13] GotoIf("SIP/4063-08298a58", "1?continue") in new stack [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Goto (macro-user-callerid,s,20) [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: GotoIf [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-user-callerid:20] NoOp("SIP/4063-08298a58", "Using CallerID "INF TEST2" <4063>") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: Noop [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: Macro [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-exten-vm:2] Set("SIP/4063-08298a58", "RingGroupMethod=none") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: Set [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-exten-vm:3] Set("SIP/4063-08298a58", "VMBOX=novm") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: Set [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-exten-vm:4] Set("SIP/4063-08298a58", "EXTTOCALL=4064") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: Set [Sep 10 10:43:03] DEBUG[1873] func_db.c: DB: CFU/4064 not found in database. [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-exten-vm:5] Set("SIP/4063-08298a58", "CFUEXT=") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: Set [Sep 10 10:43:03] DEBUG[1873] func_db.c: DB: CFB/4064 not found in database. [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-exten-vm:6] Set("SIP/4063-08298a58", "CFBEXT=") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: Set [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-exten-vm:7] Set("SIP/4063-08298a58", "RT=""") in new stack [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: Set [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-exten-vm:8] Macro("SIP/4063-08298a58", "record-enable|4064|IN") in new stack [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/4063-08298a58", "1?check") in new stack [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Goto (macro-record-enable,s,4) [Sep 10 10:43:03] DEBUG[1873] app_macro.c: Executed application: GotoIf [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Executing [s@macro-record-enable:4] AGI("SIP/4063-08298a58", "recordingcheck|20090910-104303|1252572183.0") in new stack [Sep 10 10:43:03] VERBOSE[1873] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck [Sep 10 10:43:04] VERBOSE[1873] logger.c: recordingcheck|20090910-104303|1252572183.0: Inbound recording not enabled [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- AGI Script recordingcheck completed, returning 0 [Sep 10 10:43:04] DEBUG[1873] app_macro.c: Executed application: AGI [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- Executing [s@macro-record-enable:5] MacroExit("SIP/4063-08298a58", "") in new stack [Sep 10 10:43:04] DEBUG[1873] app_macro.c: Executed application: Macro [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- Executing [s@macro-exten-vm:9] Macro("SIP/4063-08298a58", "dial||tTwW|4064") in new stack [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- Executing [s@macro-dial:1] GotoIf("SIP/4063-08298a58", "1?dial") in new stack [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- Goto (macro-dial,s,3) [Sep 10 10:43:04] DEBUG[1873] app_macro.c: Executed application: GotoIf [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- Executing [s@macro-dial:3] AGI("SIP/4063-08298a58", "dialparties.agi") in new stack [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi [Sep 10 10:43:04] VERBOSE[1873] logger.c: dialparties.agi: Starting New Dialparties.agi [Sep 10 10:43:04] VERBOSE[1878] logger.c: == Parsing '/etc/asterisk/manager.conf': [Sep 10 10:43:04] VERBOSE[1878] logger.c: Found [Sep 10 10:43:04] VERBOSE[1878] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': [Sep 10 10:43:04] VERBOSE[1878] logger.c: Found [Sep 10 10:43:04] VERBOSE[1878] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': [Sep 10 10:43:04] VERBOSE[1878] logger.c: Found [Sep 10 10:43:04] VERBOSE[1878] logger.c: == Manager 'informatica' logged on from 127.0.0.1 [Sep 10 10:43:04] VERBOSE[1873] logger.c: dialparties.agi: Caller ID name is 'INF TEST2' number is '4063' [Sep 10 10:43:04] VERBOSE[1873] logger.c: dialparties.agi: USE_CONFIRMATION: 'FALSE' [Sep 10 10:43:04] VERBOSE[1873] logger.c: dialparties.agi: RINGGROUP_INDEX: '' [Sep 10 10:43:04] VERBOSE[1873] logger.c: dialparties.agi: Methodology of ring is 'none' [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- dialparties.agi: Added extension 4064 to extension map [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- dialparties.agi: VIERI: cfignore= cidnum=4063 cf= realcalleridnum=4063 blindtransfer= [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- dialparties.agi: Extension 4064 cf is disabled [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- dialparties.agi: Extension 4064 do not disturb is disabled [Sep 10 10:43:04] VERBOSE[1873] logger.c: > dialparties.agi: extnum 4064 has: cw: 0; hascfb: 0 [] hascfu: 0 [] [Sep 10 10:43:04] VERBOSE[1873] logger.c: > dialparties.agi: ExtensionState: 0 [Sep 10 10:43:04] VERBOSE[1873] logger.c: dialparties.agi: Extension 4064 has ExtensionState: 0 [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- dialparties.agi: Checking CW and CFB status for extension 4064 [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- dialparties.agi: dbset CALLTRACE/4064 to 4063 [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- dialparties.agi: Filtered ARG3: 4064 [Sep 10 10:43:04] VERBOSE[1878] logger.c: == Manager 'informatica' logged off from 127.0.0.1 [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- AGI Script dialparties.agi completed, returning 0 [Sep 10 10:43:04] DEBUG[1873] app_macro.c: Executed application: AGI [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- Executing [s@macro-dial:7] Dial("SIP/4063-08298a58", "SIP/4064||tTwW") in new stack [Sep 10 10:43:04] DEBUG[1873] chan_sip.c: Call to peer '4064' is 1 out of 50 [Sep 10 10:43:04] VERBOSE[1873] logger.c: Audio is at 10.215.147.112 port 13430 [Sep 10 10:43:04] VERBOSE[1873] logger.c: Adding codec 0x2 (gsm) to SDP [Sep 10 10:43:04] VERBOSE[1873] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Sep 10 10:43:04] VERBOSE[1873] logger.c: Reliably Transmitting (NAT) to 10.215.146.175:5060: INVITE sip:4064@10.215.146.175:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.215.147.112:5060;branch=z9hG4bK2d026098;rport From: "INF TEST2" ;tag=as4af51617 To: Contact: Call-ID: 01c9f5bf0eeaccf63ea0a95f56cf40d0@10.215.147.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 10 Sep 2009 08:43:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 241 v=0 o=root 1412 1412 IN IP4 10.215.147.112 s=session c=IN IP4 10.215.147.112 t=0 0 m=audio 13430 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- Called 4064 [Sep 10 10:43:04] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.175:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.215.147.112:5060;branch=z9hG4bK2d026098;rport From: "INF TEST2" ;tag=as4af51617 To: Call-ID: 01c9f5bf0eeaccf63ea0a95f56cf40d0@10.215.147.112 CSeq: 102 INVITE User-Agent: Grandstream GXP280 1.2.1.4 Content-Length: 0 <-------------> [Sep 10 10:43:04] VERBOSE[1436] logger.c: --- (8 headers 0 lines) --- [Sep 10 10:43:04] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.175:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.215.147.112:5060;branch=z9hG4bK2d026098;rport From: "INF TEST2" ;tag=as4af51617 To: ;tag=bdf4a7ede2841257 Call-ID: 01c9f5bf0eeaccf63ea0a95f56cf40d0@10.215.147.112 CSeq: 102 INVITE User-Agent: Grandstream GXP280 1.2.1.4 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:04] VERBOSE[1436] logger.c: --- (10 headers 0 lines) --- [Sep 10 10:43:04] VERBOSE[1873] logger.c: -- SIP/4064-0829aa18 is ringing [Sep 10 10:43:04] VERBOSE[1873] logger.c: <--- Transmitting (NAT) to 10.215.146.165:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKe8b028b0a963e236;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Sep 10 10:43:06] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.175:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.215.147.112:5060;branch=z9hG4bK2d026098;rport From: "INF TEST2" ;tag=as4af51617 To: ;tag=bdf4a7ede2841257 Call-ID: 01c9f5bf0eeaccf63ea0a95f56cf40d0@10.215.147.112 CSeq: 102 INVITE User-Agent: Grandstream GXP280 1.2.1.4 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 214 v=0 o=4064 8000 8000 IN IP4 10.215.146.175 s=SIP Call c=IN IP4 10.215.146.175 t=0 0 m=audio 5012 RTP/AVP 3 101 a=sendrecv a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Sep 10 10:43:06] VERBOSE[1436] logger.c: --- (12 headers 11 lines) --- [Sep 10 10:43:06] WARNING[1436] chan_sip.c: PEDANTIC TESTING acked invite [Sep 10 10:43:06] VERBOSE[1436] logger.c: Found RTP audio format 3 [Sep 10 10:43:06] VERBOSE[1436] logger.c: Found RTP audio format 101 [Sep 10 10:43:06] VERBOSE[1436] logger.c: Peer audio RTP is at port 10.215.146.175:5012 [Sep 10 10:43:06] VERBOSE[1436] logger.c: Found audio description format GSM for ID 3 [Sep 10 10:43:06] VERBOSE[1436] logger.c: Found audio description format telephone-event for ID 101 [Sep 10 10:43:06] VERBOSE[1436] logger.c: Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) [Sep 10 10:43:06] VERBOSE[1436] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Sep 10 10:43:06] VERBOSE[1436] logger.c: Peer audio RTP is at port 10.215.146.175:5012 [Sep 10 10:43:06] VERBOSE[1436] logger.c: list_route: hop: [Sep 10 10:43:06] DEBUG[1436] chan_sip.c: Strict routing enforced for session 01c9f5bf0eeaccf63ea0a95f56cf40d0@10.215.147.112 [Sep 10 10:43:06] VERBOSE[1436] logger.c: set_destination: Parsing for address/port to send to [Sep 10 10:43:06] VERBOSE[1436] logger.c: set_destination: set destination to 10.215.146.175, port 5060 [Sep 10 10:43:06] VERBOSE[1436] logger.c: Transmitting (NAT) to 10.215.146.175:5060: ACK sip:4064@10.215.146.175:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.215.147.112:5060;branch=z9hG4bK38ac6b04;rport From: "INF TEST2" ;tag=as4af51617 To: ;tag=bdf4a7ede2841257 Contact: Call-ID: 01c9f5bf0eeaccf63ea0a95f56cf40d0@10.215.147.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Sep 10 10:43:06] VERBOSE[1873] logger.c: -- SIP/4064-0829aa18 answered SIP/4063-08298a58 [Sep 10 10:43:06] VERBOSE[1873] logger.c: Audio is at 10.215.147.112 port 11490 [Sep 10 10:43:06] VERBOSE[1873] logger.c: Adding codec 0x2 (gsm) to SDP [Sep 10 10:43:06] VERBOSE[1873] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Sep 10 10:43:06] VERBOSE[1873] logger.c: <--- Reliably Transmitting (NAT) to 10.215.146.165:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKe8b028b0a963e236;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=root 1412 1412 IN IP4 10.215.147.112 s=session c=IN IP4 10.215.147.112 t=0 0 m=audio 11490 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Sep 10 10:43:06] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> ACK sip:4064@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKb1d5bbbb2245773b From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:4064@10.215.147.112", nonce="09d47485", response="ca824a855fe5adf9e652de99e1ae2f2e" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 ACK User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:06] VERBOSE[1436] logger.c: --- (13 headers 0 lines) --- [Sep 10 10:43:06] WARNING[1436] chan_sip.c: PEDANTIC TESTING in find_call error 1 [Sep 10 10:43:06] WARNING[1436] chan_sip.c: PEDANTIC TESTING no call found [Sep 10 10:43:07] VERBOSE[1436] logger.c: Retransmitting #1 (NAT) to 10.215.146.165:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKe8b028b0a963e236;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=root 1412 1412 IN IP4 10.215.147.112 s=session c=IN IP4 10.215.147.112 t=0 0 m=audio 11490 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 10 10:43:07] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> ACK sip:4064@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKa2694e598852c852 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:4064@10.215.147.112", nonce="09d47485", response="ca824a855fe5adf9e652de99e1ae2f2e" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 ACK User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:07] VERBOSE[1436] logger.c: --- (13 headers 0 lines) --- [Sep 10 10:43:07] WARNING[1436] chan_sip.c: PEDANTIC TESTING in find_call error 1 [Sep 10 10:43:07] WARNING[1436] chan_sip.c: PEDANTIC TESTING no call found [Sep 10 10:43:08] VERBOSE[1436] logger.c: Retransmitting #2 (NAT) to 10.215.146.165:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKe8b028b0a963e236;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=root 1412 1412 IN IP4 10.215.147.112 s=session c=IN IP4 10.215.147.112 t=0 0 m=audio 11490 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 10 10:43:08] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> ACK sip:4064@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK04758087f287356d From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:4064@10.215.147.112", nonce="09d47485", response="ca824a855fe5adf9e652de99e1ae2f2e" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 ACK User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:08] VERBOSE[1436] logger.c: --- (13 headers 0 lines) --- [Sep 10 10:43:08] WARNING[1436] chan_sip.c: PEDANTIC TESTING in find_call error 1 [Sep 10 10:43:08] WARNING[1436] chan_sip.c: PEDANTIC TESTING no call found [Sep 10 10:43:10] VERBOSE[1436] logger.c: Retransmitting #3 (NAT) to 10.215.146.165:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKe8b028b0a963e236;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=root 1412 1412 IN IP4 10.215.147.112 s=session c=IN IP4 10.215.147.112 t=0 0 m=audio 11490 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 10 10:43:10] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> ACK sip:4064@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK35b694bce8c05360 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:4064@10.215.147.112", nonce="09d47485", response="ca824a855fe5adf9e652de99e1ae2f2e" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 ACK User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:10] VERBOSE[1436] logger.c: --- (13 headers 0 lines) --- [Sep 10 10:43:10] WARNING[1436] chan_sip.c: PEDANTIC TESTING in find_call error 1 [Sep 10 10:43:10] WARNING[1436] chan_sip.c: PEDANTIC TESTING no call found [Sep 10 10:43:14] VERBOSE[1436] logger.c: Retransmitting #4 (NAT) to 10.215.146.165:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKe8b028b0a963e236;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=root 1412 1412 IN IP4 10.215.147.112 s=session c=IN IP4 10.215.147.112 t=0 0 m=audio 11490 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 10 10:43:14] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> ACK sip:4064@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK1769b86970803e52 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:4064@10.215.147.112", nonce="09d47485", response="ca824a855fe5adf9e652de99e1ae2f2e" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 ACK User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:14] VERBOSE[1436] logger.c: --- (13 headers 0 lines) --- [Sep 10 10:43:14] WARNING[1436] chan_sip.c: PEDANTIC TESTING in find_call error 1 [Sep 10 10:43:14] WARNING[1436] chan_sip.c: PEDANTIC TESTING no call found [Sep 10 10:43:18] VERBOSE[1436] logger.c: Really destroying SIP dialog '52a6d05678408857@10.215.146.165' Method: REGISTER [Sep 10 10:43:18] VERBOSE[1436] logger.c: Retransmitting #5 (NAT) to 10.215.146.165:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKe8b028b0a963e236;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=root 1412 1412 IN IP4 10.215.147.112 s=session c=IN IP4 10.215.147.112 t=0 0 m=audio 11490 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 10 10:43:18] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> ACK sip:4064@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK164e3d57e3749f5d From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:4064@10.215.147.112", nonce="09d47485", response="ca824a855fe5adf9e652de99e1ae2f2e" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 ACK User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:18] VERBOSE[1436] logger.c: --- (13 headers 0 lines) --- [Sep 10 10:43:18] WARNING[1436] chan_sip.c: PEDANTIC TESTING in find_call error 1 [Sep 10 10:43:18] WARNING[1436] chan_sip.c: PEDANTIC TESTING no call found [Sep 10 10:43:22] VERBOSE[1914] logger.c: == Parsing '/etc/asterisk/manager.conf': [Sep 10 10:43:22] VERBOSE[1914] logger.c: Found [Sep 10 10:43:22] VERBOSE[1914] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': [Sep 10 10:43:22] VERBOSE[1914] logger.c: Found [Sep 10 10:43:22] VERBOSE[1914] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': [Sep 10 10:43:22] VERBOSE[1914] logger.c: Found [Sep 10 10:43:22] VERBOSE[1914] logger.c: == Manager 'inftester' logged on from 10.215.147.112 [Sep 10 10:43:22] VERBOSE[1914] logger.c: == Manager 'inftester' logged off from 10.215.147.112 [Sep 10 10:43:22] VERBOSE[1436] logger.c: Retransmitting #6 (NAT) to 10.215.146.165:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bKe8b028b0a963e236;received=10.215.146.165 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=root 1412 1412 IN IP4 10.215.147.112 s=session c=IN IP4 10.215.147.112 t=0 0 m=audio 11490 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 10 10:43:22] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.165:5060 ---> ACK sip:4064@10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.165:5060;branch=z9hG4bK15e346a6b9a08450 From: "TEST2" ;tag=e9445d071df0bf70 To: ;tag=as1211b496 Contact: Supported: path Proxy-Authorization: Digest username="4063", realm="asterisk", algorithm=MD5, uri="sip:4064@10.215.147.112", nonce="09d47485", response="ca824a855fe5adf9e652de99e1ae2f2e" Call-ID: 2fa63c881788d281@10.215.146.165 CSeq: 2678 ACK User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:22] VERBOSE[1436] logger.c: --- (13 headers 0 lines) --- [Sep 10 10:43:22] WARNING[1436] chan_sip.c: PEDANTIC TESTING in find_call error 1 [Sep 10 10:43:22] WARNING[1436] chan_sip.c: PEDANTIC TESTING no call found [Sep 10 10:43:26] WARNING[1436] chan_sip.c: Maximum retries exceeded on transmission 2fa63c881788d281@10.215.146.165 for seqno 2678 (Critical Response) -- See doc/sip-retransmit.txt. [Sep 10 10:43:26] WARNING[1436] chan_sip.c: Hanging up call 2fa63c881788d281@10.215.146.165 - no reply to our critical packet (see doc/sip-retransmit.txt). [Sep 10 10:43:26] VERBOSE[1873] logger.c: -- Executing [h@macro-dial:1] Macro("SIP/4063-08298a58", "hangupcall") in new stack [Sep 10 10:43:26] VERBOSE[1873] logger.c: -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/4063-08298a58", "w") in new stack [Sep 10 10:43:26] DEBUG[1873] app_macro.c: Executed application: ResetCDR [Sep 10 10:43:26] VERBOSE[1873] logger.c: -- Executing [s@macro-hangupcall:2] NoCDR("SIP/4063-08298a58", "") in new stack [Sep 10 10:43:26] DEBUG[1873] app_macro.c: Executed application: NoCDR [Sep 10 10:43:26] VERBOSE[1873] logger.c: -- Executing [s@macro-hangupcall:3] GotoIf("SIP/4063-08298a58", "1?skiprg") in new stack [Sep 10 10:43:26] VERBOSE[1873] logger.c: -- Goto (macro-hangupcall,s,6) [Sep 10 10:43:26] DEBUG[1873] app_macro.c: Executed application: GotoIf [Sep 10 10:43:26] VERBOSE[1873] logger.c: -- Executing [s@macro-hangupcall:6] GotoIf("SIP/4063-08298a58", "1?skipblkvm") in new stack [Sep 10 10:43:26] VERBOSE[1873] logger.c: -- Goto (macro-hangupcall,s,9) [Sep 10 10:43:26] DEBUG[1873] app_macro.c: Executed application: GotoIf [Sep 10 10:43:26] VERBOSE[1873] logger.c: -- Executing [s@macro-hangupcall:9] GotoIf("SIP/4063-08298a58", "1?theend") in new stack [Sep 10 10:43:26] VERBOSE[1873] logger.c: -- Goto (macro-hangupcall,s,11) [Sep 10 10:43:26] DEBUG[1873] app_macro.c: Executed application: GotoIf [Sep 10 10:43:26] VERBOSE[1873] logger.c: -- Executing [s@macro-hangupcall:11] Hangup("SIP/4063-08298a58", "") in new stack [Sep 10 10:43:26] VERBOSE[1873] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/4063-08298a58' in macro 'hangupcall' [Sep 10 10:43:26] VERBOSE[1873] logger.c: == Spawn h extension (macro-dial, h, 1) exited non-zero on 'SIP/4063-08298a58' [Sep 10 10:43:26] DEBUG[1873] chan_sip.c: Call to peer '4064' removed from call limit 50 [Sep 10 10:43:26] VERBOSE[1873] logger.c: Scheduling destruction of SIP dialog '01c9f5bf0eeaccf63ea0a95f56cf40d0@10.215.147.112' in 6400 ms (Method: INVITE) [Sep 10 10:43:26] DEBUG[1873] chan_sip.c: Strict routing enforced for session 01c9f5bf0eeaccf63ea0a95f56cf40d0@10.215.147.112 [Sep 10 10:43:26] VERBOSE[1873] logger.c: set_destination: Parsing for address/port to send to [Sep 10 10:43:26] VERBOSE[1873] logger.c: set_destination: set destination to 10.215.146.175, port 5060 [Sep 10 10:43:26] VERBOSE[1873] logger.c: Reliably Transmitting (NAT) to 10.215.146.175:5060: BYE sip:4064@10.215.146.175:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.215.147.112:5060;branch=z9hG4bK460cadf9;rport From: "INF TEST2" ;tag=as4af51617 To: ;tag=bdf4a7ede2841257 Call-ID: 01c9f5bf0eeaccf63ea0a95f56cf40d0@10.215.147.112 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Sep 10 10:43:26] VERBOSE[1873] logger.c: == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/4063-08298a58' in macro 'dial' [Sep 10 10:43:26] VERBOSE[1873] logger.c: == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/4063-08298a58' in macro 'exten-vm' [Sep 10 10:43:26] VERBOSE[1873] logger.c: == Spawn extension (from-internal, 4064, 1) exited non-zero on 'SIP/4063-08298a58' [Sep 10 10:43:26] DEBUG[1873] chan_sip.c: Call from peer '4063' removed from call limit 50 [Sep 10 10:43:26] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.175:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.215.147.112:5060;branch=z9hG4bK460cadf9;rport From: "INF TEST2" ;tag=as4af51617 To: ;tag=bdf4a7ede2841257 Call-ID: 01c9f5bf0eeaccf63ea0a95f56cf40d0@10.215.147.112 CSeq: 103 BYE User-Agent: Grandstream GXP280 1.2.1.4 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Sep 10 10:43:26] VERBOSE[1436] logger.c: --- (11 headers 0 lines) --- [Sep 10 10:43:28] VERBOSE[1436] logger.c: Really destroying SIP dialog '01c9f5bf0eeaccf63ea0a95f56cf40d0@10.215.147.112' Method: INVITE [Sep 10 10:43:28] VERBOSE[1436] logger.c: Really destroying SIP dialog '2fa63c881788d281@10.215.146.165' Method: INVITE [Sep 10 10:43:34] VERBOSE[1862] logger.c: -- Remote UNIX connection disconnected [Sep 10 10:43:37] ERROR[1444] chan_iax2.c: Call rejected, CallToken Support required. If unexpected, resolve by placing address 127.0.0.1 in the calltokenignore list or setting user 10006 requirecalltoken=no [Sep 10 10:43:37] ERROR[1443] chan_iax2.c: Call rejected, CallToken Support required. If unexpected, resolve by placing address 127.0.0.1 in the calltokenignore list or setting user 10000 requirecalltoken=no [Sep 10 10:43:37] ERROR[1442] chan_iax2.c: Call rejected, CallToken Support required. If unexpected, resolve by placing address 127.0.0.1 in the calltokenignore list or setting user 10005 requirecalltoken=no [Sep 10 10:43:37] ERROR[1447] chan_iax2.c: Call rejected, CallToken Support required. If unexpected, resolve by placing address 127.0.0.1 in the calltokenignore list or setting user 10003 requirecalltoken=no [Sep 10 10:43:38] ERROR[1441] chan_iax2.c: Call rejected, CallToken Support required. If unexpected, resolve by placing address 127.0.0.1 in the calltokenignore list or setting user 10001 requirecalltoken=no [Sep 10 10:43:38] ERROR[1444] chan_iax2.c: Call rejected, CallToken Support required. If unexpected, resolve by placing address 127.0.0.1 in the calltokenignore list or setting user 10004 requirecalltoken=no [Sep 10 10:43:38] ERROR[1443] chan_iax2.c: Call rejected, CallToken Support required. If unexpected, resolve by placing address 127.0.0.1 in the calltokenignore list or setting user 10007 requirecalltoken=no [Sep 10 10:43:38] ERROR[1442] chan_iax2.c: Call rejected, CallToken Support required. If unexpected, resolve by placing address 127.0.0.1 in the calltokenignore list or setting user 10002 requirecalltoken=no [Sep 10 10:43:42] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.175:5060 ---> REGISTER sip:10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.175:5060;branch=z9hG4bKf6aab5aa77b1f093 From: "TEST" ;tag=633f68e304023661 To: Contact: ;reg-id=1;+sip.instance="" Supported: path Authorization: Digest username="4064", realm="asterisk", algorithm=MD5, uri="sip:10.215.147.112", nonce="5d04e769", response="8e5e78c57eaf26f13f201f4a8f908bfb" Call-ID: 5f67912771b0c4c7@10.215.146.175 CSeq: 10007 REGISTER Expires: 120 User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:42] VERBOSE[1436] logger.c: --- (14 headers 0 lines) --- [Sep 10 10:43:42] VERBOSE[1436] logger.c: Using latest REGISTER request as basis request [Sep 10 10:43:42] VERBOSE[1436] logger.c: Sending to 10.215.146.175 : 5060 (no NAT) [Sep 10 10:43:42] VERBOSE[1436] logger.c: <--- Transmitting (NAT) to 10.215.146.175:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.215.146.175:5060;branch=z9hG4bKf6aab5aa77b1f093;received=10.215.146.175 From: "TEST" ;tag=633f68e304023661 To: Call-ID: 5f67912771b0c4c7@10.215.146.175 CSeq: 10007 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Sep 10 10:43:42] VERBOSE[1436] logger.c: <--- Transmitting (NAT) to 10.215.146.175:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.215.146.175:5060;branch=z9hG4bKf6aab5aa77b1f093;received=10.215.146.175 From: "TEST" ;tag=633f68e304023661 To: ;tag=as13bf58ba Call-ID: 5f67912771b0c4c7@10.215.146.175 CSeq: 10007 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="25d10d51" Content-Length: 0 <------------> [Sep 10 10:43:42] VERBOSE[1436] logger.c: Scheduling destruction of SIP dialog '5f67912771b0c4c7@10.215.146.175' in 32000 ms (Method: REGISTER) [Sep 10 10:43:42] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.175:5060 ---> REGISTER sip:10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.175:5060;branch=z9hG4bKbccf1a390e46753f From: "TEST" ;tag=633f68e304023661 To: Contact: ;reg-id=1;+sip.instance="" Supported: path Authorization: Digest username="4064", realm="asterisk", algorithm=MD5, uri="sip:10.215.147.112", nonce="25d10d51", response="1e1daa621e99caad551fb489d96f3dd9" Call-ID: 5f67912771b0c4c7@10.215.146.175 CSeq: 10008 REGISTER Expires: 120 User-Agent: Grandstream GXP280 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Sep 10 10:43:42] VERBOSE[1436] logger.c: --- (14 headers 0 lines) --- [Sep 10 10:43:42] VERBOSE[1436] logger.c: Using latest REGISTER request as basis request [Sep 10 10:43:42] VERBOSE[1436] logger.c: Sending to 10.215.146.175 : 5060 (NAT) [Sep 10 10:43:42] VERBOSE[1436] logger.c: <--- Transmitting (NAT) to 10.215.146.175:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.215.146.175:5060;branch=z9hG4bKbccf1a390e46753f;received=10.215.146.175 From: "TEST" ;tag=633f68e304023661 To: Call-ID: 5f67912771b0c4c7@10.215.146.175 CSeq: 10008 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Sep 10 10:43:42] VERBOSE[1436] logger.c: Reliably Transmitting (NAT) to 10.215.146.175:5060: OPTIONS sip:4064@10.215.146.175:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.215.147.112:5060;branch=z9hG4bK4d1e0308;rport From: "Unknown" ;tag=as2ab4ee84 To: Contact: Call-ID: 3536d6b93923e05325d4b5ec2d49ae7c@10.215.147.112 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 10 Sep 2009 08:43:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- [Sep 10 10:43:42] VERBOSE[1436] logger.c: <--- Transmitting (NAT) to 10.215.146.175:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.215.146.175:5060;branch=z9hG4bKbccf1a390e46753f;received=10.215.146.175 From: "TEST" ;tag=633f68e304023661 To: ;tag=as13bf58ba Call-ID: 5f67912771b0c4c7@10.215.146.175 CSeq: 10008 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Expires: 120 Contact: ;expires=120 Date: Thu, 10 Sep 2009 08:43:42 GMT Content-Length: 0 <------------> [Sep 10 10:43:42] VERBOSE[1436] logger.c: Scheduling destruction of SIP dialog '5f67912771b0c4c7@10.215.146.175' in 32000 ms (Method: REGISTER) [Sep 10 10:43:42] VERBOSE[1436] logger.c: <--- SIP read from 10.215.146.175:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.215.147.112:5060;branch=z9hG4bK4d1e0308;rport From: "Unknown" ;tag=as2ab4ee84 To: ;tag=bdf4a7ede2841257 Call-ID: 3536d6b93923e05325d4b5ec2d49ae7c@10.215.147.112 CSeq: 102 OPTIONS User-Agent: Grandstream GXP280 1.2.1.4 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Sep 10 10:43:42] VERBOSE[1436] logger.c: --- (11 headers 0 lines) --- [Sep 10 10:43:42] VERBOSE[1436] logger.c: Really destroying SIP dialog '3536d6b93923e05325d4b5ec2d49ae7c@10.215.147.112' Method: OPTIONS