<-------------> asterisk*CLI> <--- SIP read from UDP://10.0.0.41:5060 ---> INVITE sip:950956141@10.0.0.4 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.41:5060;branch=z9hG4bK-d5d4ff0f From: "Operadora" ;tag=4f64cbaaf0ac9557o0 To: Call-ID: 58564c66-126cabcb@10.0.0.41 CSeq: 101 INVITE Max-Forwards: 70 Contact: "Operadora" Expires: 240 User-Agent: Cisco/SPA525G2-7.4.7 Content-Length: 306 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 63621754 63621754 IN IP4 10.0.0.41 s=- c=IN IP4 10.0.0.41 t=0 0 m=audio 16458 RTP/AVP 18 0 2 8 9 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (14 headers 15 lines) --- == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Sending to 10.0.0.41 : 5060 (no NAT) Using INVITE request as basis request - 58564c66-126cabcb@10.0.0.41 Found user '100' for '100' <--- Reliably Transmitting (NAT) to 10.0.0.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.41:5060;branch=z9hG4bK-d5d4ff0f;received=10.0.0.41 From: "Operadora" ;tag=4f64cbaaf0ac9557o0 To: ;tag=as54e8c8cf Call-ID: 58564c66-126cabcb@10.0.0.41 CSeq: 101 INVITE User-Agent: Asterisk PBX 1.6.0.28 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timerP/VozTelecom1450/950956141 extension prueba@from-next WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="70164d16" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '58564c66-126cabcb@10.0.0.41' in 32000 ms (Method: INVITE) asterisk*CLI> <--- SIP read from UDP://10.0.0.41:5060 ---> ACK sip:950956141@10.0.0.4 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.41:5060;branch=z9hG4bK-d5d4ff0f From: "Operadora" ;tag=4f64cbaaf0ac9557o0 To: ;tag=as54e8c8cf Call-ID: 58564c66-126cabcb@10.0.0.41 CSeq: 101 ACK Max-Forwards: 70 Contact: "Operadora" User-Agent: Cisco/SPA525G2-7.4.7 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk*CLI> <--- SIP read from UDP://10.0.0.41:5060 ---> INVITE sip:950956141@10.0.0.4 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.41:5060;branch=z9hG4bK-9c80fd23 From: "Operadora" ;tag=4f64cbaaf0ac9557o0 To: Call-ID: 58564c66-126cabcb@10.0.0.41 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="100",realm="asterisk",nonce="70164d16",uri="sip:950956141@10.0.0.4",algorithm=MD5,response="d5d56d7a193ad02865c5ddc98c0e55b7" Contact: "Operadora" Expires: 240 User-Agent: Cisco/SPA525G2-7.4.7 Content-Length: 306 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 63621754 63621754 IN IP4 10.0.0.41 s=- c=IN IP4 10.0.0.41 t=0 0 m=audio 16458 RTP/AVP 18 0 2 8 9 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (15 headers 15 lines) --- Sending to 10.0.0.41 : 5060 (NAT) Using INVITE request as basis request - 58564c66-126cabcb@10.0.0.41 Found user '100' for '100' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 101 Found audio description format G729a for ID 18 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format PCMA for ID 8 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x190c (ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.0.0.41:16458 Looking for 950956141 in from-next (domain 10.0.0.4) list_route: hop: 0/950956141 extension prueba@from-next <--- Transmitting (NAT) to 10.0.0.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.41:5060;branch=z9hG4bK-9c80fd23;received=10.0.0.41 From: "Operadora" ;tag=4f64cbaaf0ac9557o0 To: Call-ID: 58564c66-126cabcb@10.0.0.41 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.28 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [950956141@from-next:1] NoOp("SIP/100-000004bd", "") in new stack -- Executing [950956141@from-next:2] SIPAddHeader("SIP/100-000004bd", "Remote-Party-ID:\;privacy=off\;party=calling") in new stack -- Executing [950956141@from-next:3] Dial("SIP/100-000004bd", "SIP/NEXTC/950956141,,r") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 192.168.5.100 port 17794om1450/950956141 extension prueba@from-next Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 193.22.119.20:5060: INVITE sip:950956141@voztele.com SIP/2.0 Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK024d3ccd;rport Max-Forwards: 70 From: "device" ;tag=as4bcfb12a To: Contact: Call-ID: 369063023bae2c7c6be05561032b6659@voztele.com CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.28 Date: Wed, 27 Apr 2011 16:12:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Remote-Party-ID: ;privacy=off;party=calling Content-Type: application/sdp Content-Length: 313 v=0 o=root 1834020316 1834020316 IN IP4 192.168.5.100 s=Asterisk PBX 1.6.0.28 c=IN IP4 192.168.5.100 t=0 0 m=audio 17794 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called NEXTC/950956141 <--- Transmitting (NAT) to 10.0.0.41:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.41:5060;branch=z9hG4bK-9c80fd23;received=10.0.0.41 From: "Operadora" ;tag=4f64cbaaf0ac9557o0 To: ;tag=as24f25188 Call-ID: 58564c66-126cabcb@10.0.0.41 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.28 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> asterisk*CLI> <--- SIP read from UDP://193.22.119.20:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK024d3ccd;rport=5060;received=217.18.231.97 From: "device" ;tag=as4bcfb12a To: ;tag=ba3d51acad53eeb51d56ab2459dbff7b.1ed9 Call-ID: 369063023bae2c7c6be05561032b6659@voztele.com CSeq: 102 INVITE Proxy-Authenticate: Digest realm="voztele.com", nonce="1112" Server: OpenSER (1.2.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Transmitting (NAT) to 193.22.119.20:5060: ACK sip:950956141@voztele.com SIP/2.0 Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK024d3ccd;rport Max-Forwards: 70 From: "device" ;tag=as4bcfb12a To: ;tag=ba3d51acad53eeb51d56ab2459dbff7b.1ed9 Contact: Call-ID: 369063023bae2c7c6be05561032b6659@voztele.com CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.28 Content-Length: 0 --- Audio is at 192.168.5.100 port 17794 Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 193.22.119.20:5060: INVITE sip:950956141@voztele.com SIP/2.0 Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK4c742655;rport Max-Forwards: 70 From: "device" ;tag=as4bcfb12a To: Contact: Call-ID: 369063023bae2c7c6be05561032b6659@voztele.com CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.0.28 Proxy-Authorization: Digest username="3400023628", realm="voztele.com", algorithm=MD5, uri="sip:950956141@voztele.com", nonce="4444", response="222" Date: Wed, 27 Apr 2011 16:12:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Remote-Party-ID: ;privacy=off;party=calling Content-Type: application/sdp Content-Length: 313 v=0 o=root 1834020316 1834020317 IN IP4 192.168.5.100 s=Asterisk PBX 1.6.0.28 c=IN IP4 192.168.5.100 t=0 0 m=audio 17794 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #1 (NAT) to 192.168.31.82:5060:956141 OPTIONS sip:55555116@192.168.1.5 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK7d354227;rport Max-Forwards: 70 From: "Unknown" ;tag=as30fcac36 To: Contact: Call-ID: 33d7630a360acd2e2bdc491624c065b0@10.0.0.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.28 Date: Wed, 27 Apr 2011 16:12:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> 1 <--- SIP read from UDP://192.168.31.82:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK7d354227;rport From: "Unknown" ;tag=as30fcac36 To: ;tag=f62f599f81c971ef Call-ID: 33d7630a360acd2e2bdc491624c065b0@10.0.0.4 CSeq: 102 OPTIONS User-Agent: Grandstream HT287 1.1.0.42 DevId 000b8227712b Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Supported: replaces, timer Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '33d7630a360acd2e2bdc491624c065b0@10.0.0.4' Method: OPTIONS asterisk*CLI> <--- SIP read from UDP://192.168.31.101:5060 ---> <-------------> asterisk*CLI> <--- SIP read from UDP://193.22.119.20:5060 ---> SIP/2.0 100 Runing CPL script... Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK4c742655;rport=5060;received=217.18.231.97 From: "device" ;tag=as4bcfb12a To: Call-ID: 369063023bae2c7c6be05561032b6659@voztele.com CSeq: 103 INVITE Server: OpenSER (1.2.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk*CLI> <--- SIP read from UDP://193.22.119.20:5060 ---> SIP/2.0 100 Runing CPL script... Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK4c742655;rport=5060;received=217.18.231.97 From: "device" ;tag=as4bcfb12a To: Call-ID: 369063023bae2c7c6be05561032b6659@voztele.com CSeq: 103 INVITE Server: OpenSER (1.2.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk*CLI> <--- SIP read from UDP://193.22.119.20:5060 ---> INVITE sip:s@192.168.5.100 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKeebe.514c9214.0 Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKeebe.414c9214.0 Via: SIP/2.0/UDP 192.168.5.100:5060;received=217.18.231.97;branch=z9hG4bK4c742655;rport=5060 Max-Forwards: 31 From: "device" ;tag=as4bcfb12a To: Contact: Call-ID: 369063023bae2c7c6be05561032b6659@voztele.com CSeq: 103 INVITE Date: Wed, 27 Apr 2011 16:12:20 GMT Remote-Party-ID: ;privacy=off;party=calling Content-Type: application/sdp Content-Length: 331 P-RTP-Proxy: YES P-Asserted-Identity: X-CPLFROM: 3400023550 v=0 o=root 1834020316 1834020317 IN IP4 192.168.5.100 s=Asterisk PBX 1.6.0.28 c=IN IP4 193.22.119.14 t=0 0 m=audio 58764 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes <-------------> --- (19 headers 15 lines) --- <--- Transmitting (NAT) to 193.22.119.20:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKeebe.514c9214.0;received=193.22.119.20 Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKeebe.414c9214.0 Via: SIP/2.0/UDP 192.168.5.100:5060;received=217.18.231.97;branch=z9hG4bK4c742655;rport=5060 Record-Route: Record-Route: From: "device" ;tag=as4bcfb12a To: Call-ID: 369063023bae2c7c6be05561032b6659@voztele.com CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.0.28 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Now forwarding SIP/100-000004bd to 'Local/s@from-voztele' (thanks to SIP/NEXTC-000004be) Scheduling destruction of SIP dialog '369063023bae2c7c6be05561032b6659@voztele.com' in 6400 ms (Method: INVITE) [Apr 27 18:12:20] WARNING[17947]: chan_sip.c:15559 func_header_read: This function can only be used on SIP channels. -- Executing [s@from-voztele:1] NoOp("Local/s@from-voztele-6197;2", "") in new stack Reliably Transmitting (NAT) to 193.22.119.20:5060: CANCEL sip:950956141@voztele.com SIP/2.0 Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK4c742655;rport Max-Forwards: 70 From: "device" ;tag=as4bcfb12a To: Call-ID: 369063023bae2c7c6be05561032b6659@voztele.com CSeq: 103 CANCEL User-Agent: Asterisk PBX 1.6.0.28 ontent-Length: 0 --- [Apr 27 18:12:20] WARNING[17947]: chan_sip.c:15559 func_header_read: This function can only be used on SIP channels. Scheduling destruction of SIP dialog '369063023bae2c7c6be05561032b6659@voztele.com' in 6400 ms (Method: INVITE) -- Executing [s@from-voztele:2] Set("Local/s@from-voztele-6197;2", "dialed=") in new stack -- Executing [s@from-voztele:3] Goto("Local/s@from-voztele-6197;2", "ext-did,,1") in new stack -- Goto (ext-did,s,1) [Apr 27 18:12:20] WARNING[17947]: pbx.c:3781 __ast_pbx_run: Channel 'Local/s@from-voztele-6197;2' sent into invalid extension 's' in context 'ext-did', but no invalid handler == Everyone is busy/congested at this time (1:0/0/1)xtension prueba@from-next -- Executing [950956141@from-next:4] Hangup("SIP/100-000004bd", "") in new stack == Spawn extension (from-next, 950956141, 4) exited non-zero on 'SIP/100-000004bd' Scheduling destruction of SIP dialog '58564c66-126cabcb@10.0.0.41' in 32000 ms (Method: INVITE) asterisk*CLI> <--- Reliably Transmitting (NAT) to 10.0.0.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 10.0.0.41:5060;branch=z9hG4bK-9c80fd23;received=10.0.0.41 From: "Operadora" ;tag=4f64cbaaf0ac9557o0 To: ;tag=as24f25188 Call-ID: 58564c66-126cabcb@10.0.0.41 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.28 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> asterisk*CLI> <--- SIP read from UDP://10.0.0.41:5060 ---> ACK sip:950956141@10.0.0.4 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.41:5060;branch=z9hG4bK-9c80fd23 From: "Operadora" ;tag=4f64cbaaf0ac9557o0 To: ;tag=as24f25188 Call-ID: 58564c66-126cabcb@10.0.0.41 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="100",realm="asterisk",nonce="70164d16",uri="sip:950956141@10.0.0.4",algorithm=MD5,response="d5d56d7a193ad02865c5ddc98c0e55b7" Contact: "Operadora" User-Agent: Cisco/SPA525G2-7.4.7 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Reliably Transmitting (NAT) to 192.168.3.61:10000:41 extension prueba@from-next OPTIONS sip:55555105@192.168.3.61:10000 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK36723d93;rport Max-Forwards: 70 From: "Unknown" ;tag=as463ca7e5 To: Contact: Call-ID: 3ab91c9b23ba5d2149c83355303e1d25@10.0.0.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.28 Date: Wed, 27 Apr 2011 16:12:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> <--- SIP read from UDP://193.22.119.20:5060 ---> SIP/2.0 200 canceling Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK4c742655;rport=5060;received=217.18.231.97 From: "device" ;tag=as4bcfb12a To: ;tag=0dc66d454840fa7d0b618c9f0fdb83b7-4fd7 Call-ID: 369063023bae2c7c6be05561032b6659@voztele.com CSeq: 103 CANCEL Server: OpenSER (1.2.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk*CLI> <--- SIP read from UDP://193.22.119.20:5060 ---> CANCEL sip:s@192.168.5.100 SIP/2.0 Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKeebe.514c9214.0 Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKeebe.414c9214.0 Via: SIP/2.0/UDP 192.168.5.100:5060;received=217.18.231.97;branch=z9hG4bK4c742655;rport=5060 Max-Forwards: 31 From: "device" ;tag=as4bcfb12a To: Call-ID: 369063023bae2c7c6be05561032b6659@voztele.com CSeq: 103 CANCEL Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 193.22.119.20 : 5060 (NAT) Scheduling destruction of SIP dialog '369063023bae2c7c6be05561032b6659@voztele.com' in 6400 ms (Method: CANCEL) <--- Reliably Transmitting (NAT) to 193.22.119.20:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK024d3ccd;received=193.22.119.20;rport=5060 From: "device" ;tag=as4bcfb12a To: ;tag=0dc66d454840fa7d0b618c9f0fdb83b7-4fd7 Call-ID: 369063023bae2c7c6be05561032b6659@voztele.com CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.28 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> <--- Transmitting (NAT) to 193.22.119.20:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKeebe.514c9214.0;received=193.22.119.20 Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKeebe.414c9214.0 Via: SIP/2.0/UDP 192.168.5.100:5060;received=217.18.231.97;branch=z9hG4bK4c742655;rport=5060 From: "device" ;tag=as4bcfb12a To: ;tag=0dc66d454840fa7d0b618c9f0fdb83b7-4fd7 Call-ID: 369063023bae2c7c6be05561032b6659@voztele.com CSeq: 103 CANCEL User-Agent: Asterisk PBX 1.6.0.28 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> asterisk*CLI> <--- SIP read from UDP://192.168.3.61:10000 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK36723d93;rport From: "Unknown" ;tag=as463ca7e5 To: ;tag=2bf206200cf25dd2 Call-ID: 3ab91c9b23ba5d2149c83355303e1d25@10.0.0.4 CSeq: 102 OPTIONS User-Agent: Grandstream HT287 1.1.0.42 DevId 000b8225b682 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Supported: replaces, timer Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '3ab91c9b23ba5d2149c83355303e1d25@10.0.0.4' Method: OPTIONS Retransmitting #1 (NAT) to 193.22.119.20:5060:956141 extension prueba@from-next SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK024d3ccd;received=193.22.119.20;rport=5060 From: "device" ;tag=as4bcfb12a To: ;tag=0dc66d454840fa7d0b618c9f0fdb83b7-4fd7 Call-ID: 369063023bae2c7c6be05561032b6659@voztele.com CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.28 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (NAT) to 193.22.119.20:5060:956141 extension prueba@from-next SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK024d3ccd;received=193.22.119.20;rport=5060 From: "device" ;tag=as4bcfb12a To: ;tag=0dc66d454840fa7d0b618c9f0fdb83b7-4fd7 Call-ID: 369063023bae2c7c6be05561032b6659@voztele.com CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.28 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0