Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:38:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:56329 [Aug 8 09:38:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411515@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:38:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411485@127.0.0.1' [Aug 8 09:38:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411485@127.0.0.1 [Aug 8 09:38:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411485@127.0.0.1' Method: OPTIONS [Aug 8 09:38:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:38:45] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:41701 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411525@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411525@127.0.0.1 [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:38:45] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:38:45] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:38:45] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411525@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:38:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:38:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:38:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:38:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:38:45] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:38:45] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:41701 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=41701 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as3a47f570 Call-ID: 1344411525@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:38:45] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:41701 [Aug 8 09:38:45] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411525@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:38:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411495@127.0.0.1' [Aug 8 09:38:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411495@127.0.0.1 [Aug 8 09:38:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411495@127.0.0.1' Method: OPTIONS [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2052 ---> INVITE sip:2210@192.168.0.178 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-efllbwvm696a;rport From: "2209" ;tag=gbdfdvpkud To: Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v CSeq: 1 INVITE Max-Forwards: 70 Contact: ;reg-id=1 X-Serialnumber: 000413341D08 P-Key-Flags: keys="3" User-Agent: snom300/8.4.32 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 331 v=0 o=root 1987219225 1987219225 IN IP4 192.168.1.102 s=call c=IN IP4 192.168.1.102 t=0 0 m=audio 11460 RTP/SAVP 8 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:JATbmi5O79v44WmAN8Q/Ad5xGjCkgnmjDLf5IdvF a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 0 [ 37]: INVITE sip:2210@192.168.0.178 SIP/2.0 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-efllbwvm696a;rport [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 2 [ 52]: From: "2209" ;tag=gbdfdvpkud [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 3 [ 28]: To: [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 4 [ 34]: Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 7 [ 61]: Contact: ;reg-id=1 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 8 [ 28]: X-Serialnumber: 000413341D08 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 9 [ 21]: P-Key-Flags: keys="3" [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 10 [ 26]: User-Agent: snom300/8.4.32 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 11 [ 23]: Accept: application/sdp [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 12 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 13 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 14 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 15 [ 35]: Session-Expires: 3600;refresher=uas [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 16 [ 10]: Min-SE: 90 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 17 [ 29]: Content-Type: application/sdp [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 18 [ 19]: Content-Length: 331 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 19 [ 0]: [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 0 [ 3]: v=0 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 1 [ 49]: o=root 1987219225 1987219225 IN IP4 192.168.1.102 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 2 [ 6]: s=call [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.1.102 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 5 [ 30]: m=audio 11460 RTP/SAVP 8 0 101 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:JATbmi5O79v44WmAN8Q/Ad5xGjCkgnmjDLf5IdvF [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 12 [ 10]: a=sendrecv [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: --- (19 headers 13 lines) --- [Aug 8 09:38:48] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:38:48] DEBUG[2371] acl.c: For destination '192.168.1.102', our source address is '192.168.0.178'. [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.0.178:5060 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 3c39db8cc1a5-pv4xf5eh5w4v - INVITE (No RTP) [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Aug 8 09:38:48] DEBUG[2371] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change" [Aug 8 09:38:48] DEBUG[2371] sip/reqresp_parser.c: Found SIP option: -timer- [Aug 8 09:38:48] DEBUG[2371] sip/reqresp_parser.c: Matched SIP option: timer [Aug 8 09:38:48] DEBUG[2371] sip/reqresp_parser.c: Found SIP option: -100rel- [Aug 8 09:38:48] DEBUG[2371] sip/reqresp_parser.c: Matched SIP option: 100rel [Aug 8 09:38:48] DEBUG[2371] sip/reqresp_parser.c: Found SIP option: -replaces- [Aug 8 09:38:48] DEBUG[2371] sip/reqresp_parser.c: Matched SIP option: replaces [Aug 8 09:38:48] DEBUG[2371] sip/reqresp_parser.c: Found SIP option: -from-change- [Aug 8 09:38:48] DEBUG[2371] sip/reqresp_parser.c: Matched SIP option: from-change [Aug 8 09:38:48] DEBUG[2371] netsock2.c: Splitting '192.168.1.102:2052' into... [Aug 8 09:38:48] DEBUG[2371] netsock2.c: ...host '192.168.1.102' and port '2052'. [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Sending to 192.168.1.102:2052 (NAT) [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Initializing initreq for method INVITE - callid 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Using INVITE request as basis request - 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:38:48] DEBUG[2371] netsock2.c: Splitting '192.168.0.178' into... [Aug 8 09:38:48] DEBUG[2371] netsock2.c: ...host '192.168.0.178' and port ''. [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Found peer '2209' for '2209' from 192.168.1.102:2052 [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.1.102:2052 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-efllbwvm696a;received=192.168.1.102;rport=2052 From: "2209" ;tag=gbdfdvpkud To: ;tag=as49009171 Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v CSeq: 1 INVITE Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="tvox", nonce="39ce2b62" Content-Length: 0 <------------> [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #286 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.102:2052 [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '3c39db8cc1a5-pv4xf5eh5w4v' in 32000 ms (Method: INVITE) [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2052 ---> ACK sip:2210@192.168.0.178 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-efllbwvm696a;rport From: "2209" ;tag=gbdfdvpkud To: ;tag=as49009171 Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v CSeq: 1 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 0 [ 34]: ACK sip:2210@192.168.0.178 SIP/2.0 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-efllbwvm696a;rport [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 2 [ 52]: From: "2209" ;tag=gbdfdvpkud [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 3 [ 43]: To: ;tag=as49009171 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 4 [ 34]: Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 7 [ 61]: Contact: ;reg-id=1 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 9 [ 0]: [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: --- (9 headers 0 lines) --- [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #286 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Stopping retransmission on '3c39db8cc1a5-pv4xf5eh5w4v' of Response 1: Match Found [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2052 ---> INVITE sip:2210@192.168.0.178 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-ny4ij1tsds9k;rport From: "2209" ;tag=gbdfdvpkud To: Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v CSeq: 2 INVITE Max-Forwards: 70 Contact: ;reg-id=1 X-Serialnumber: 000413341D08 P-Key-Flags: keys="3" User-Agent: snom300/8.4.32 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Authorization: Digest username="2209",realm="tvox",nonce="39ce2b62",uri="sip:2210@192.168.0.178",response="d1a86c8c531981cf4b8f20408d97fb3c",algorithm=MD5 Content-Type: application/sdp Content-Length: 331 v=0 o=root 1987219225 1987219225 IN IP4 192.168.1.102 s=call c=IN IP4 192.168.1.102 t=0 0 m=audio 11460 RTP/SAVP 8 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:JATbmi5O79v44WmAN8Q/Ad5xGjCkgnmjDLf5IdvF a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 0 [ 37]: INVITE sip:2210@192.168.0.178 SIP/2.0 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-ny4ij1tsds9k;rport [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 2 [ 52]: From: "2209" ;tag=gbdfdvpkud [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 3 [ 28]: To: [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 4 [ 34]: Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 5 [ 14]: CSeq: 2 INVITE [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 7 [ 61]: Contact: ;reg-id=1 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 8 [ 28]: X-Serialnumber: 000413341D08 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 9 [ 21]: P-Key-Flags: keys="3" [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 10 [ 26]: User-Agent: snom300/8.4.32 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 11 [ 23]: Accept: application/sdp [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 12 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 13 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 14 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 15 [ 35]: Session-Expires: 3600;refresher=uas [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 16 [ 10]: Min-SE: 90 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 17 [154]: Authorization: Digest username="2209",realm="tvox",nonce="39ce2b62",uri="sip:2210@192.168.0.178",response="d1a86c8c531981cf4b8f20408d97fb3c",algorithm=MD5 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 18 [ 29]: Content-Type: application/sdp [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 19 [ 19]: Content-Length: 331 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 20 [ 0]: [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 0 [ 3]: v=0 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 1 [ 49]: o=root 1987219225 1987219225 IN IP4 192.168.1.102 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 2 [ 6]: s=call [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.1.102 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 5 [ 30]: m=audio 11460 RTP/SAVP 8 0 101 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:JATbmi5O79v44WmAN8Q/Ad5xGjCkgnmjDLf5IdvF [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Body 12 [ 10]: a=sendrecv [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: --- (20 headers 13 lines) --- [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Aug 8 09:38:48] DEBUG[2371] netsock2.c: Splitting '192.168.1.102:2052' into... [Aug 8 09:38:48] DEBUG[2371] netsock2.c: ...host '192.168.1.102' and port '2052'. [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Sending to 192.168.1.102:2052 (NAT) [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Initializing initreq for method INVITE - callid 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Using INVITE request as basis request - 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:38:48] DEBUG[2371] netsock2.c: Splitting '192.168.0.178' into... [Aug 8 09:38:48] DEBUG[2371] netsock2.c: ...host '192.168.0.178' and port ''. [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Found peer '2209' for '2209' from 192.168.1.102:2052 [Aug 8 09:38:48] DEBUG[2371] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x1c3a7dc8' [Aug 8 09:38:48] DEBUG[2371] res_rtp_asterisk.c: Allocated port 18588 for RTP instance '0x1c3a7dc8' [Aug 8 09:38:48] DEBUG[2371] rtp_engine.c: RTP instance '0x1c3a7dc8' is setup and ready to go [Aug 8 09:38:48] DEBUG[2371] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x1c3a7dc8' [Aug 8 09:38:48] VERBOSE[2371] netsock2.c: == Using SIP RTP TOS bits 184 [Aug 8 09:38:48] VERBOSE[2371] netsock2.c: == Using SIP RTP CoS mark 5 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Setting NAT on RTP to On [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Processing session-level SDP o=root 1987219225 1987219225 IN IP4 192.168.1.102... UNSUPPORTED OR FAILED. [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED OR FAILED. [Aug 8 09:38:48] DEBUG[2371] netsock2.c: Splitting '192.168.1.102' into... [Aug 8 09:38:48] DEBUG[2371] netsock2.c: ...host '192.168.1.102' and port ''. [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.102... OK. [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Found RTP audio format 8 [Aug 8 09:38:48] DEBUG[2371] rtp_engine.c: Setting payload 8 based on m type on 0x41f0b5b0 [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Found RTP audio format 0 [Aug 8 09:38:48] DEBUG[2371] rtp_engine.c: Setting payload 0 based on m type on 0x41f0b5b0 [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Found RTP audio format 101 [Aug 8 09:38:48] DEBUG[2371] rtp_engine.c: Setting payload 101 based on m type on 0x41f0b5b0 [Aug 8 09:38:48] DEBUG[2371] sip/sdp_crypto.c: local_key64 CStJb7zHpMiDirfLQN85uIIeE/fGvo9zKi69GowG len 40 [Aug 8 09:38:48] DEBUG[2371] res_srtp.c: Adding new policy for SSRC 1420768450 [Aug 8 09:38:48] DEBUG[2371] sip/sdp_crypto.c: SRTP policy activated [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:JATbmi5O79v44WmAN8Q/Ad5xGjCkgnmjDLf5IdvF... OK. [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 8 09:38:48] DEBUG[2371] rtp_engine.c: Incorporating payload 0 on 0x41f0b5b0 [Aug 8 09:38:48] DEBUG[2371] rtp_engine.c: Incorporating payload 8 on 0x41f0b5b0 [Aug 8 09:38:48] DEBUG[2371] rtp_engine.c: Incorporating payload 101 on 0x41f0b5b0 [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 8 09:38:48] DEBUG[2371] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c3a7dc8' [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Peer audio RTP is at port 192.168.1.102:11460 [Aug 8 09:38:48] DEBUG[2371] rtp_engine.c: Copying payload 0 from 0x41f0b5b0 to 0x1c3a7f90 [Aug 8 09:38:48] DEBUG[2371] rtp_engine.c: Copying payload 8 from 0x41f0b5b0 to 0x1c3a7f90 [Aug 8 09:38:48] DEBUG[2371] rtp_engine.c: Copying payload 101 from 0x41f0b5b0 to 0x1c3a7f90 [Aug 8 09:38:48] DEBUG[2371] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x1c3a7dc8' [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Checking SIP call limits for device 2209 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Updating call counter for incoming call [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Call from peer '2209' is 1 out of 2147483647 [Aug 8 09:38:48] DEBUG[2371] netsock2.c: Splitting '192.168.0.178' into... [Aug 8 09:38:48] DEBUG[2371] netsock2.c: ...host '192.168.0.178' and port ''. [Aug 8 09:38:48] DEBUG[2371] netsock2.c: Splitting '192.168.0.178' into... [Aug 8 09:38:48] DEBUG[2371] netsock2.c: ...host '192.168.0.178' and port ''. [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: Looking for 2210 in test_issue (domain 192.168.0.178) [Aug 8 09:38:48] DEBUG[2988] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/2209-00000000 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 2209 CallerIDName: Unknown AccountCode: Exten: 2210 Context: test_issue Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: *** Our native formats are 0x8 (alaw) [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: This channel will not be able to handle video. [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: list_route: hop: [Aug 8 09:38:48] DEBUG[2982] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/2209-00000000 Uniqueid: 1344411528.0 Channeltype: SIP SIPcallid: 3c39db8cc1a5-pv4xf5eh5w4v SIPfullcontact: sip:2209@192.168.1.102:2052;line=882y5m72 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/2209-00000000 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 2209 CallerIDName: Unknown AccountCode: Exten: 2210 Context: test_issue Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: SIPURI Value: sip:2209@192.168.1.102:2052 Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: SIPDOMAIN Value: 192.168.0.178 Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: SIPCALLID Value: 3c39db8cc1a5-pv4xf5eh5w4v Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2209 [Aug 8 09:38:48] DEBUG[2321] chan_sip.c: Checking device state for peer 2209 [Aug 8 09:38:48] DEBUG[2321] devicestate.c: Changing state for SIP/2209 - state 2 (In use) [Aug 8 09:38:48] DEBUG[2321] devicestate.c: device 'SIP/2209' state '2' [Aug 8 09:38:48] DEBUG[2980] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/2209-00000000 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 2209 CallerIDName: Unknown AccountCode: Exten: 2210 Context: test_issue Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Session-Expires: 3600 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Refresher: UAS [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Received Min-SE: 90 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Session timer started: 288 - 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: SIP/2209-00000000: New call is still down.... Trying... [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.102:2052 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-ny4ij1tsds9k;received=192.168.1.102;rport=2052 From: "2209" ;tag=gbdfdvpkud To: Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v CSeq: 2 INVITE Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.1.102:2052 [Aug 8 09:38:48] DEBUG[2980] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2209-00000000 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[2988] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2209-00000000 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2209-00000000 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[5008] pbx.c: Result of 'EXTEN' is '2210' [Aug 8 09:38:48] DEBUG[5008] pbx.c: Launching 'Dial' [Aug 8 09:38:48] VERBOSE[5008] pbx.c: -- Executing [2210@test_issue:1] Dial("SIP/2209-00000000", "SIP/2210") in new stack [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/2209-00000000 Context: test_issue Extension: 2210 Priority: 1 Application: Dial AppData: SIP/2210 Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[2984] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/2209-00000000 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 2209 CallerIDName: Unknown AccountCode: Exten: 2210 Context: test_issue Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[2984] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2209-00000000 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[2988] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2209 Context: telenia_localextensions Hint: SIP/2209 Status: 1 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2209 Context: telenia_localextensions Hint: SIP/2209 Status: 1 [Aug 8 09:38:48] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2209 [Aug 8 09:38:48] DEBUG[2321] chan_sip.c: Checking device state for peer 2209 [Aug 8 09:38:48] DEBUG[2321] devicestate.c: Changing state for SIP/2209 - state 2 (In use) [Aug 8 09:38:48] DEBUG[2321] devicestate.c: device 'SIP/2209' state '2' [Aug 8 09:38:48] DEBUG[2980] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2209 Context: telenia_localextensions Hint: SIP/2209 Status: 1 [Aug 8 09:38:48] DEBUG[2984] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2209 Context: telenia_localextensions Hint: SIP/2209 Status: 1 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: DIALSTATUS Value: Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: DIALEDPEERNUMBER Value: Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: DIALEDPEERNAME Value: Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: ANSWEREDTIME Value: Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: DIALEDTIME Value: Uniqueid: 1344411528.0 [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Allocating new SIP dialog for 30afb97e347d97f540829244214bb6e8@(null) - INVITE (No RTP) [Aug 8 09:38:48] DEBUG[5008] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x1c3bfc88' [Aug 8 09:38:48] DEBUG[5008] res_rtp_asterisk.c: Allocated port 10866 for RTP instance '0x1c3bfc88' [Aug 8 09:38:48] DEBUG[5008] rtp_engine.c: RTP instance '0x1c3bfc88' is setup and ready to go [Aug 8 09:38:48] DEBUG[5008] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x1c3bfc88' [Aug 8 09:38:48] VERBOSE[5008] netsock2.c: == Using SIP RTP TOS bits 184 [Aug 8 09:38:48] VERBOSE[5008] netsock2.c: == Using SIP RTP CoS mark 5 [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Setting NAT on RTP to On [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 8 09:38:48] DEBUG[5008] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:38:48] DEBUG[5008] acl.c: For destination '192.168.2.210', our source address is '192.168.0.178'. [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.0.178:5060 [Aug 8 09:38:48] DEBUG[2980] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/2210-00000001 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: Unknown AccountCode: Exten: Context: test_issue Uniqueid: 1344411528.1 [Aug 8 09:38:48] DEBUG[2984] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/2210-00000001 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: Unknown AccountCode: Exten: Context: test_issue Uniqueid: 1344411528.1 [Aug 8 09:38:48] DEBUG[2988] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/2210-00000001 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: Unknown AccountCode: Exten: Context: test_issue Uniqueid: 1344411528.1 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/2210-00000001 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: Unknown AccountCode: Exten: Context: test_issue Uniqueid: 1344411528.1 [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: *** Our native formats are 0x8 (alaw) [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: This channel will not be able to handle video. [Aug 8 09:38:48] DEBUG[5008] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 8 09:38:48] DEBUG[5008] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: SIPCALLID Value: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 Uniqueid: 1344411528.1 [Aug 8 09:38:48] DEBUG[2982] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 Channeltype: SIP SIPcallid: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 SIPfullcontact: sip:2210@192.168.2.210:2048;line=nh5ckpq1 [Aug 8 09:38:48] DEBUG[2982] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/2210-00000001 Channeltype: SIP SIPcallid: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 SIPfullcontact: sip:2210@192.168.2.210:2048;line=nh5ckpq1 Peername: 2210 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: DIALEDPEERNUMBER Value: 2210 Uniqueid: 1344411528.1 [Aug 8 09:38:48] DEBUG[5008] rtp_engine.c: Seeded SDP of 'SIP/2210-00000001' with that of 'SIP/2209-00000000' [Aug 8 09:38:48] DEBUG[5008] channel.c: Not copying variable DIALEDTIME. [Aug 8 09:38:48] DEBUG[5008] channel.c: Not copying variable ANSWEREDTIME. [Aug 8 09:38:48] DEBUG[5008] channel.c: Not copying variable DIALEDPEERNAME. [Aug 8 09:38:48] DEBUG[5008] channel.c: Not copying variable DIALEDPEERNUMBER. [Aug 8 09:38:48] DEBUG[5008] channel.c: Not copying variable DIALSTATUS. [Aug 8 09:38:48] DEBUG[5008] channel.c: Not copying variable SIPCALLID. [Aug 8 09:38:48] DEBUG[5008] channel.c: Not copying variable SIPDOMAIN. [Aug 8 09:38:48] DEBUG[5008] channel.c: Not copying variable SIPURI. [Aug 8 09:38:48] DEBUG[2980] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/2210-00000001 CallerIDNum: 2210 CallerIDName: Uniqueid: 1344411528.1 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Aug 8 09:38:48] DEBUG[2984] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/2210-00000001 CallerIDNum: 2210 CallerIDName: Uniqueid: 1344411528.1 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Aug 8 09:38:48] DEBUG[2988] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/2210-00000001 CallerIDNum: 2210 CallerIDName: Uniqueid: 1344411528.1 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/2210-00000001 CallerIDNum: 2210 CallerIDName: Uniqueid: 1344411528.1 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Direct media not possible when using SRTP, ignoring canreinvite setting [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Outgoing Call for 2210 [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Updating call counter for outgoing call [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Call to peer '2210' is 1 out of 2147483647 [Aug 8 09:38:48] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2210 [Aug 8 09:38:48] DEBUG[2321] chan_sip.c: Checking device state for peer 2210 [Aug 8 09:38:48] DEBUG[2321] devicestate.c: Changing state for SIP/2210 - state 6 (Ringing) [Aug 8 09:38:48] DEBUG[2321] devicestate.c: device 'SIP/2210' state '6' [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Aug 8 09:38:48] VERBOSE[5008] chan_sip.c: Audio is at 10866 [Aug 8 09:38:48] DEBUG[5008] sip/sdp_crypto.c: local_key64 rrG/MleANMxuoCfjtPyp5qmmiZmZVJdBKyWgS1yP len 40 [Aug 8 09:38:48] VERBOSE[5008] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 8 09:38:48] VERBOSE[5008] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 8 09:38:48] VERBOSE[5008] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: -- Done with adding codecs to SDP [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Initializing initreq for method INVITE - callid 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Header 0 [ 56]: INVITE sip:2210@192.168.2.210:2048;line=nh5ckpq1 SIP/2.0 [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK2f3ffc93;rport [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Header 3 [ 55]: From: "Unknown" ;tag=as0fa7c2f6 [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Header 4 [ 47]: To: [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Header 5 [ 38]: Contact: [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Header 6 [ 60]: Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:38:48 GMT [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 8 09:38:48] VERBOSE[5008] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: INVITE sip:2210@192.168.2.210:2048;line=nh5ckpq1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK2f3ffc93;rport Max-Forwards: 70 From: "Unknown" ;tag=as0fa7c2f6 To: Contact: Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 102 INVITE User-Agent: asterisk Date: Wed, 08 Aug 2012 07:38:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 331 v=0 o=tvox 70257168 70257168 IN IP4 192.168.0.178 s=asterisk c=IN IP4 192.168.0.178 t=0 0 m=audio 10866 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:rrG/MleANMxuoCfjtPyp5qmmiZmZVJdBKyWgS1yP --- [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #290 [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.2.210:2048 [Aug 8 09:38:48] DEBUG[2980] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/2209-00000000 Destination: SIP/2210-00000001 CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: UniqueID: 1344411528.0 DestUniqueID: 1344411528.1 Dialstring: 2210 [Aug 8 09:38:48] DEBUG[2984] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/2209-00000000 Destination: SIP/2210-00000001 CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: UniqueID: 1344411528.0 DestUniqueID: 1344411528.1 Dialstring: 2210 [Aug 8 09:38:48] DEBUG[2988] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/2209-00000000 Destination: SIP/2210-00000001 CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: UniqueID: 1344411528.0 DestUniqueID: 1344411528.1 Dialstring: 2210 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/2209-00000000 Destination: SIP/2210-00000001 CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: UniqueID: 1344411528.0 DestUniqueID: 1344411528.1 Dialstring: 2210 [Aug 8 09:38:48] VERBOSE[5008] app_dial.c: -- Called SIP/2210 [Aug 8 09:38:48] DEBUG[2323] app_queue.c: Extension '2209@telenia_localextensions' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 8 09:38:48] DEBUG[2980] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 8 [Aug 8 09:38:48] DEBUG[2984] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 8 [Aug 8 09:38:48] DEBUG[2988] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 8 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 8 [Aug 8 09:38:48] DEBUG[2323] app_queue.c: Extension '2210@telenia_localextensions' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Aug 8 09:38:48] DEBUG[2394] app_queue.c: Device 'SIP/2209' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 8 09:38:48] DEBUG[2394] app_queue.c: Device 'SIP/2209' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 8 09:38:48] DEBUG[2394] app_queue.c: Device 'SIP/2210' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK2f3ffc93;rport=5060 From: "Unknown" ;tag=as0fa7c2f6 To: ;tag=1del7f6fcr Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 102 INVITE Contact: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK2f3ffc93;rport=5060 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 2 [ 55]: From: "Unknown" ;tag=as0fa7c2f6 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 3 [ 62]: To: ;tag=1del7f6fcr [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 4 [ 60]: Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 8 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: Header 10 [ 0]: [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: --- (10 headers 0 lines) --- [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: *** SIP TIMER: Cancelling retransmission #290 - INVITE (got response) [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060' Request 102: Found [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: SIP response 180 to standard invite [Aug 8 09:38:48] DEBUG[2371] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Aug 8 09:38:48] VERBOSE[2371] chan_sip.c: list_route: hop: [Aug 8 09:38:48] DEBUG[2988] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2210-00000001 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 2210 CallerIDName: ConnectedLineNum: 2209 ConnectedLineName: Unknown Uniqueid: 1344411528.1 [Aug 8 09:38:48] DEBUG[3176] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2210-00000001 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 2210 CallerIDName: ConnectedLineNum: 2209 ConnectedLineName: Unknown Uniqueid: 1344411528.1 [Aug 8 09:38:48] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2210 [Aug 8 09:38:48] DEBUG[2321] chan_sip.c: Checking device state for peer 2210 [Aug 8 09:38:48] DEBUG[2321] devicestate.c: Changing state for SIP/2210 - state 6 (Ringing) [Aug 8 09:38:48] DEBUG[2321] devicestate.c: device 'SIP/2210' state '6' [Aug 8 09:38:48] VERBOSE[5008] app_dial.c: -- SIP/2210-00000001 is ringing [Aug 8 09:38:48] DEBUG[5008] rtp_engine.c: Setting early bridge SDP of 'SIP/2209-00000000' with that of 'SIP/2210-00000001' [Aug 8 09:38:48] VERBOSE[5008] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.102:2052 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-ny4ij1tsds9k;received=192.168.1.102;rport=2052 From: "2209" ;tag=gbdfdvpkud To: ;tag=as0bbc9cb2 Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v CSeq: 2 INVITE Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Aug 8 09:38:48] DEBUG[5008] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.1.102:2052 [Aug 8 09:38:48] DEBUG[2980] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2210-00000001 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 2210 CallerIDName: ConnectedLineNum: 2209 ConnectedLineName: Unknown Uniqueid: 1344411528.1 [Aug 8 09:38:48] DEBUG[2984] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2210-00000001 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 2210 CallerIDName: ConnectedLineNum: 2209 ConnectedLineName: Unknown Uniqueid: 1344411528.1 [Aug 8 09:38:48] DEBUG[2394] app_queue.c: Device 'SIP/2210' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Aug 8 09:38:49] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK2f3ffc93;rport=5060 From: "Unknown" ;tag=as0fa7c2f6 To: ;tag=1del7f6fcr Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 102 INVITE Contact: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Aug 8 09:38:49] DEBUG[2371] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Aug 8 09:38:49] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK2f3ffc93;rport=5060 [Aug 8 09:38:49] DEBUG[2371] chan_sip.c: Header 2 [ 55]: From: "Unknown" ;tag=as0fa7c2f6 [Aug 8 09:38:49] DEBUG[2371] chan_sip.c: Header 3 [ 62]: To: ;tag=1del7f6fcr [Aug 8 09:38:49] DEBUG[2371] chan_sip.c: Header 4 [ 60]: Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:38:49] DEBUG[2371] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 8 09:38:49] DEBUG[2371] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 8 09:38:49] DEBUG[2371] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 8 09:38:49] DEBUG[2371] chan_sip.c: Header 8 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 8 09:38:49] DEBUG[2371] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 8 09:38:49] DEBUG[2371] chan_sip.c: Header 10 [ 0]: [Aug 8 09:38:49] VERBOSE[2371] chan_sip.c: --- (10 headers 0 lines) --- [Aug 8 09:38:49] DEBUG[2371] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060' Request 102: Found [Aug 8 09:38:49] DEBUG[2371] chan_sip.c: SIP response 180 to standard invite [Aug 8 09:38:49] DEBUG[2371] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Aug 8 09:38:49] VERBOSE[2371] chan_sip.c: list_route: hop: [Aug 8 09:38:49] VERBOSE[5008] app_dial.c: -- SIP/2210-00000001 is ringing [Aug 8 09:38:49] DEBUG[5008] rtp_engine.c: Setting early bridge SDP of 'SIP/2209-00000000' with that of 'SIP/2210-00000001' [Aug 8 09:38:51] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK2f3ffc93;rport=5060 From: "Unknown" ;tag=as0fa7c2f6 To: ;tag=1del7f6fcr Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 102 INVITE Contact: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Aug 8 09:38:51] DEBUG[2371] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Aug 8 09:38:51] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK2f3ffc93;rport=5060 [Aug 8 09:38:51] DEBUG[2371] chan_sip.c: Header 2 [ 55]: From: "Unknown" ;tag=as0fa7c2f6 [Aug 8 09:38:51] DEBUG[2371] chan_sip.c: Header 3 [ 62]: To: ;tag=1del7f6fcr [Aug 8 09:38:51] DEBUG[2371] chan_sip.c: Header 4 [ 60]: Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:38:51] DEBUG[2371] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 8 09:38:51] DEBUG[2371] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 8 09:38:51] DEBUG[2371] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 8 09:38:51] DEBUG[2371] chan_sip.c: Header 8 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 8 09:38:51] DEBUG[2371] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 8 09:38:51] DEBUG[2371] chan_sip.c: Header 10 [ 0]: [Aug 8 09:38:51] VERBOSE[2371] chan_sip.c: --- (10 headers 0 lines) --- [Aug 8 09:38:51] DEBUG[2371] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060' Request 102: Found [Aug 8 09:38:51] DEBUG[2371] chan_sip.c: SIP response 180 to standard invite [Aug 8 09:38:51] DEBUG[2371] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Aug 8 09:38:51] VERBOSE[2371] chan_sip.c: list_route: hop: [Aug 8 09:38:51] VERBOSE[5008] app_dial.c: -- SIP/2210-00000001 is ringing [Aug 8 09:38:51] DEBUG[5008] rtp_engine.c: Setting early bridge SDP of 'SIP/2209-00000000' with that of 'SIP/2210-00000001' [Aug 8 09:38:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:38:54] DEBUG[5008] res_rtp_asterisk.c: RTCP NAT: Got RTCP from other end. Now sending to address 192.168.2.210:11987 [Aug 8 09:38:54] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 82 bytes [Aug 8 09:38:54] DEBUG[5008] res_rtp_asterisk.c: RTCP Read too short [Aug 8 09:38:54] DEBUG[5008] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c3bfc88' [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK2f3ffc93;rport=5060 From: "Unknown" ;tag=as0fa7c2f6 To: ;tag=1del7f6fcr Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 102 INVITE Contact: ;reg-id=1 User-Agent: snom300/8.4.32 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 329 v=0 o=root 172176250 172176251 IN IP4 192.168.2.210 s=call c=IN IP4 192.168.2.210 t=0 0 m=audio 11986 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:t/gyZ4pOluHvmshU2RCkno3w2gc93Ca/lz3lLjVQ a=sendrecv <-------------> [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK2f3ffc93;rport=5060 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 2 [ 55]: From: "Unknown" ;tag=as0fa7c2f6 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 3 [ 62]: To: ;tag=1del7f6fcr [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 4 [ 60]: Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 8 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 9 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 10 [ 40]: Supported: 100rel, replaces, from-change [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 12 [ 19]: Content-Length: 329 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Body 0 [ 3]: v=0 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Body 1 [ 47]: o=root 172176250 172176251 IN IP4 192.168.2.210 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Body 2 [ 6]: s=call [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.2.210 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Body 5 [ 30]: m=audio 11986 RTP/SAVP 8 0 101 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Body 11 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:t/gyZ4pOluHvmshU2RCkno3w2gc93Ca/lz3lLjVQ [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Body 12 [ 10]: a=sendrecv [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: --- (13 headers 13 lines) --- [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Acked pending invite 102 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Stopping retransmission on '690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060' of Request 102: Match Found [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: SIP response 200 to standard invite [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Processing session-level SDP o=root 172176250 172176251 IN IP4 192.168.2.210... UNSUPPORTED OR FAILED. [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED OR FAILED. [Aug 8 09:38:54] DEBUG[2371] netsock2.c: Splitting '192.168.2.210' into... [Aug 8 09:38:54] DEBUG[2371] netsock2.c: ...host '192.168.2.210' and port ''. [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.2.210... OK. [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: Found RTP audio format 8 [Aug 8 09:38:54] DEBUG[2371] rtp_engine.c: Setting payload 8 based on m type on 0x41f0ad00 [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: Found RTP audio format 0 [Aug 8 09:38:54] DEBUG[2371] rtp_engine.c: Setting payload 0 based on m type on 0x41f0ad00 [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: Found RTP audio format 101 [Aug 8 09:38:54] DEBUG[2371] rtp_engine.c: Setting payload 101 based on m type on 0x41f0ad00 [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Aug 8 09:38:54] DEBUG[2371] res_srtp.c: Adding new policy for SSRC 797210039 [Aug 8 09:38:54] DEBUG[2371] sip/sdp_crypto.c: SRTP policy activated [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:t/gyZ4pOluHvmshU2RCkno3w2gc93Ca/lz3lLjVQ... OK. [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 8 09:38:54] DEBUG[2371] rtp_engine.c: Incorporating payload 0 on 0x41f0ad00 [Aug 8 09:38:54] DEBUG[2371] rtp_engine.c: Incorporating payload 8 on 0x41f0ad00 [Aug 8 09:38:54] DEBUG[2371] rtp_engine.c: Incorporating payload 101 on 0x41f0ad00 [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 8 09:38:54] DEBUG[2371] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c3bfc88' [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: Peer audio RTP is at port 192.168.2.210:11986 [Aug 8 09:38:54] DEBUG[2371] rtp_engine.c: Copying payload 0 from 0x41f0ad00 to 0x1c3bfe50 [Aug 8 09:38:54] DEBUG[2371] rtp_engine.c: Copying payload 8 from 0x41f0ad00 to 0x1c3bfe50 [Aug 8 09:38:54] DEBUG[2371] rtp_engine.c: Copying payload 101 from 0x41f0ad00 to 0x1c3bfe50 [Aug 8 09:38:54] DEBUG[2371] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x1c3bfc88' [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: We have an owner, now see if we need to change this call [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Updating call counter for outgoing call [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: list_route: hop: [Aug 8 09:38:54] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2210 [Aug 8 09:38:54] DEBUG[2321] chan_sip.c: Checking device state for peer 2210 [Aug 8 09:38:54] DEBUG[2321] devicestate.c: Changing state for SIP/2210 - state 2 (In use) [Aug 8 09:38:54] DEBUG[2321] devicestate.c: device 'SIP/2210' state '2' [Aug 8 09:38:54] DEBUG[2982] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/2210-00000001 Channeltype: SIP Uniqueid: 1344411528.1 SIPcallid: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 SIPfullcontact: sip:2210@192.168.2.210:2048;line=nh5ckpq1 Peername: 2210 [Aug 8 09:38:54] DEBUG[2323] app_queue.c: Extension '2210@telenia_localextensions' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 8 09:38:54] DEBUG[2394] app_queue.c: Device 'SIP/2210' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 8 09:38:54] DEBUG[2980] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 1 [Aug 8 09:38:54] DEBUG[2984] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 1 [Aug 8 09:38:54] DEBUG[2988] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 1 [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 1 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Strict routing enforced for session 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 8 09:38:54] DEBUG[2371] netsock2.c: Splitting '192.168.2.210:2048' into... [Aug 8 09:38:54] DEBUG[2371] netsock2.c: ...host '192.168.2.210' and port '2048'. [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: set_destination: set destination to 192.168.2.210:2048 [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: Transmitting (NAT) to 192.168.2.210:2048: ACK sip:2210@192.168.2.210:2048;line=nh5ckpq1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK3ac58c39;rport Max-Forwards: 70 From: "Unknown" ;tag=as0fa7c2f6 To: ;tag=1del7f6fcr Contact: Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 102 ACK User-Agent: asterisk Content-Length: 0 --- [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Trying to put 'ACK sip:221' onto UDP socket destined for 192.168.2.210:2048 [Aug 8 09:38:54] VERBOSE[5008] app_dial.c: -- SIP/2210-00000001 answered SIP/2209-00000000 [Aug 8 09:38:54] DEBUG[5008] rtp_engine.c: Setting early bridge SDP of 'SIP/2209-00000000' with that of 'SIP/2210-00000001' [Aug 8 09:38:54] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2210 [Aug 8 09:38:54] DEBUG[2321] chan_sip.c: Checking device state for peer 2210 [Aug 8 09:38:54] DEBUG[2321] devicestate.c: Changing state for SIP/2210 - state 2 (In use) [Aug 8 09:38:54] DEBUG[2321] devicestate.c: device 'SIP/2210' state '2' [Aug 8 09:38:54] DEBUG[2980] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2210-00000001 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2210 CallerIDName: ConnectedLineNum: 2209 ConnectedLineName: Unknown Uniqueid: 1344411528.1 [Aug 8 09:38:54] DEBUG[2984] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2210-00000001 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2210 CallerIDName: ConnectedLineNum: 2209 ConnectedLineName: Unknown Uniqueid: 1344411528.1 [Aug 8 09:38:54] DEBUG[2988] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2210-00000001 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2210 CallerIDName: ConnectedLineNum: 2209 ConnectedLineName: Unknown Uniqueid: 1344411528.1 [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2210-00000001 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2210 CallerIDName: ConnectedLineNum: 2209 ConnectedLineName: Unknown Uniqueid: 1344411528.1 [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: DIALSTATUS Value: ANSWER Uniqueid: 1344411528.0 [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: DIALEDPEERNAME Value: SIP/2210-00000001 Uniqueid: 1344411528.0 [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: DIALEDPEERNUMBER Value: 2210 Uniqueid: 1344411528.0 [Aug 8 09:38:54] DEBUG[2394] app_queue.c: Device 'SIP/2210' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPEER Value: SIP/2210-00000001 Uniqueid: 1344411528.0 [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPEER Value: SIP/2209-00000000 Uniqueid: 1344411528.1 [Aug 8 09:38:54] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2209 [Aug 8 09:38:54] DEBUG[2321] chan_sip.c: Checking device state for peer 2209 [Aug 8 09:38:54] DEBUG[2321] devicestate.c: Changing state for SIP/2209 - state 2 (In use) [Aug 8 09:38:54] DEBUG[2321] devicestate.c: device 'SIP/2209' state '2' [Aug 8 09:38:54] DEBUG[2394] app_queue.c: Device 'SIP/2209' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 8 09:38:54] DEBUG[2984] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2209-00000000 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: Unknown Uniqueid: 1344411528.0 [Aug 8 09:38:54] DEBUG[2988] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2209-00000000 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: Unknown Uniqueid: 1344411528.0 [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2209-00000000 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: Unknown Uniqueid: 1344411528.0 [Aug 8 09:38:54] DEBUG[5008] chan_sip.c: SIP answering channel: SIP/2209-00000000 [Aug 8 09:38:54] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:38:54] DEBUG[5008] chan_sip.c: Setting framing from config on incoming call [Aug 8 09:38:54] DEBUG[5008] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 8 09:38:54] DEBUG[5008] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Aug 8 09:38:54] VERBOSE[5008] chan_sip.c: Audio is at 18588 [Aug 8 09:38:54] VERBOSE[5008] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 8 09:38:54] VERBOSE[5008] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 8 09:38:54] VERBOSE[5008] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 8 09:38:54] DEBUG[5008] chan_sip.c: -- Done with adding codecs to SDP [Aug 8 09:38:54] DEBUG[5008] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 8 09:38:54] VERBOSE[5008] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.1.102:2052 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-ny4ij1tsds9k;received=192.168.1.102;rport=2052 From: "2209" ;tag=gbdfdvpkud To: ;tag=as0bbc9cb2 Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v CSeq: 2 INVITE Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 333 v=0 o=tvox 842690090 842690090 IN IP4 192.168.0.178 s=asterisk c=IN IP4 192.168.0.178 t=0 0 m=audio 18588 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:CStJb7zHpMiDirfLQN85uIIeE/fGvo9zKi69GowG <------------> [Aug 8 09:38:54] DEBUG[5008] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #293 [Aug 8 09:38:54] DEBUG[5008] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.102:2052 [Aug 8 09:38:54] DEBUG[2984] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 AccountCode: OldAccountCode: [Aug 8 09:38:54] DEBUG[2988] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 AccountCode: OldAccountCode: [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 AccountCode: OldAccountCode: [Aug 8 09:38:54] DEBUG[5008] features.c: bridge answer set, chan answer set [Aug 8 09:38:54] DEBUG[5008] features.c: Removing dialed interfaces datastore on SIP/2210-00000001 since we're bridging [Aug 8 09:38:54] DEBUG[2984] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:38:54] DEBUG[2988] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2052 ---> ACK sip:2210@192.168.0.178:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-5d5anaonv9yr;rport From: "2209" ;tag=gbdfdvpkud To: ;tag=as0bbc9cb2 Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v CSeq: 2 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 0 [ 39]: ACK sip:2210@192.168.0.178:5060 SIP/2.0 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-5d5anaonv9yr;rport [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 2 [ 52]: From: "2209" ;tag=gbdfdvpkud [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 3 [ 43]: To: ;tag=as0bbc9cb2 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 4 [ 34]: Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 5 [ 11]: CSeq: 2 ACK [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 7 [ 61]: Contact: ;reg-id=1 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Header 9 [ 0]: [Aug 8 09:38:54] VERBOSE[2371] chan_sip.c: --- (9 headers 0 lines) --- [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #293 [Aug 8 09:38:54] DEBUG[2371] chan_sip.c: Stopping retransmission on '3c39db8cc1a5-pv4xf5eh5w4v' of Response 2: Match Found [Aug 8 09:38:54] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:38:54] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPEER Value: SIP/2210-00000001 Uniqueid: 1344411528.0 [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPVTCALLID Value: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 Uniqueid: 1344411528.0 [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPEER Value: SIP/2209-00000000 Uniqueid: 1344411528.1 [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPVTCALLID Value: 3c39db8cc1a5-pv4xf5eh5w4v Uniqueid: 1344411528.1 [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPEER Value: SIP/2210-00000001 Uniqueid: 1344411528.0 [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPVTCALLID Value: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 Uniqueid: 1344411528.0 [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPEER Value: SIP/2209-00000000 Uniqueid: 1344411528.1 [Aug 8 09:38:54] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPVTCALLID Value: 3c39db8cc1a5-pv4xf5eh5w4v Uniqueid: 1344411528.1 [Aug 8 09:38:54] DEBUG[2980] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2209-00000000 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: Unknown Uniqueid: 1344411528.0 [Aug 8 09:38:54] DEBUG[2980] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 AccountCode: OldAccountCode: [Aug 8 09:38:54] DEBUG[2980] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:38:54] DEBUG[5008] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Aug 8 09:38:54] DEBUG[5008] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Aug 8 09:38:54] DEBUG[5008] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x1c3a7dc8' [Aug 8 09:38:54] DEBUG[5008] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Aug 8 09:38:54] DEBUG[5008] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Aug 8 09:38:55] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:36167 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411535@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411535@127.0.0.1 [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:38:55] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:38:55] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:38:55] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411535@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:38:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:38:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:38:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:38:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:38:55] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:38:55] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:36167 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=36167 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as02c5b394 Call-ID: 1344411535@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:38:55] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:36167 [Aug 8 09:38:55] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411535@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:38:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411505@127.0.0.1' [Aug 8 09:38:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411505@127.0.0.1 [Aug 8 09:38:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411505@127.0.0.1' Method: OPTIONS [Aug 8 09:38:59] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:38:59] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:39:04] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:04] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:58970 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411545@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411545@127.0.0.1 [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:39:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:39:05] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:39:05] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411545@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:39:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:39:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:39:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:39:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:39:05] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:39:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:58970 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=58970 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as7516314a Call-ID: 1344411545@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:39:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:58970 [Aug 8 09:39:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411545@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:39:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411515@127.0.0.1' [Aug 8 09:39:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411515@127.0.0.1 [Aug 8 09:39:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411515@127.0.0.1' Method: OPTIONS [Aug 8 09:39:09] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:09] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:39:14] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:14] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:15] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:47769 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411555@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411555@127.0.0.1 [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:39:15] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:39:15] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:39:15] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411555@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:39:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:39:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:39:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:39:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:39:15] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:39:15] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:47769 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=47769 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as55028342 Call-ID: 1344411555@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:39:15] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:47769 [Aug 8 09:39:15] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411555@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:39:17] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411525@127.0.0.1' [Aug 8 09:39:17] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411525@127.0.0.1 [Aug 8 09:39:17] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411525@127.0.0.1' Method: OPTIONS [Aug 8 09:39:19] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:19] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:39:24] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:24] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:25] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:45857 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411565@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411565@127.0.0.1 [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:39:25] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:39:25] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:39:25] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411565@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:39:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:39:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:39:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:39:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:39:25] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:39:25] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:45857 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=45857 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as76629395 Call-ID: 1344411565@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:39:25] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:45857 [Aug 8 09:39:25] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411565@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:39:27] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411535@127.0.0.1' [Aug 8 09:39:27] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411535@127.0.0.1 [Aug 8 09:39:27] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411535@127.0.0.1' Method: OPTIONS [Aug 8 09:39:29] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:29] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 0286ea34023073b16d77c9626e52ab4a@(null) - OPTIONS (No RTP) [Aug 8 09:39:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:39:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 76c541b00ca2d7166c39fe4337c03131@127.0.0.1:5060 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK648d3c92 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as689328b4 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 76c541b00ca2d7166c39fe4337c03131@127.0.0.1:5060 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:39:32 GMT [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:39:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK648d3c92 Max-Forwards: 70 From: "Unknown" ;tag=as689328b4 To: Contact: Call-ID: 76c541b00ca2d7166c39fe4337c03131@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:39:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #300 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:39:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK648d3c92 Max-Forwards: 70 From: "Unknown" ;tag=as689328b4 To: Contact: Call-ID: 76c541b00ca2d7166c39fe4337c03131@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:39:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK648d3c92 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as689328b4 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 76c541b00ca2d7166c39fe4337c03131@127.0.0.1:5060 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:39:32 GMT [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:39:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:39:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:39:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:39:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:39:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:39:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:39:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK648d3c92;received=127.0.0.1 From: "Unknown" ;tag=as689328b4 To: ;tag=as689328b4 Call-ID: 76c541b00ca2d7166c39fe4337c03131@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:39:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '76c541b00ca2d7166c39fe4337c03131@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:39:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK648d3c92;received=127.0.0.1 From: "Unknown" ;tag=as689328b4 To: ;tag=as689328b4 Call-ID: 76c541b00ca2d7166c39fe4337c03131@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK648d3c92;received=127.0.0.1 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as689328b4 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as689328b4 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 76c541b00ca2d7166c39fe4337c03131@127.0.0.1:5060 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:39:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #300 [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '76c541b00ca2d7166c39fe4337c03131@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:39:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 76c541b00ca2d7166c39fe4337c03131@127.0.0.1:5060 [Aug 8 09:39:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '76c541b00ca2d7166c39fe4337c03131@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:39:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:39:34] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:34] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:35] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:45995 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411575@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411575@127.0.0.1 [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:39:35] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:39:35] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:39:35] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411575@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:39:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:39:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:39:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:39:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:39:35] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:39:35] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:45995 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=45995 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as243863c2 Call-ID: 1344411575@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:39:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:45995 [Aug 8 09:39:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411575@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:39:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411545@127.0.0.1' [Aug 8 09:39:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411545@127.0.0.1 [Aug 8 09:39:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411545@127.0.0.1' Method: OPTIONS [Aug 8 09:39:39] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:39] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:39:44] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:44] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:45] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:34490 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411585@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411585@127.0.0.1 [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:39:45] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:39:45] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:39:45] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411585@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:39:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:39:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:39:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:39:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:39:45] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:39:45] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:34490 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=34490 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as5d5ba2e1 Call-ID: 1344411585@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:39:45] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:34490 [Aug 8 09:39:45] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411585@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:39:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411555@127.0.0.1' [Aug 8 09:39:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411555@127.0.0.1 [Aug 8 09:39:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411555@127.0.0.1' Method: OPTIONS [Aug 8 09:39:49] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:49] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:39:54] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:54] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:55] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:38224 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411595@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411595@127.0.0.1 [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:39:55] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:39:55] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:39:55] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411595@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:39:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:39:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:39:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:39:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:39:55] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:39:55] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:38224 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=38224 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as29c77f92 Call-ID: 1344411595@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:39:55] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:38224 [Aug 8 09:39:55] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411595@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:39:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411565@127.0.0.1' [Aug 8 09:39:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411565@127.0.0.1 [Aug 8 09:39:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411565@127.0.0.1' Method: OPTIONS [Aug 8 09:39:59] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:39:59] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:40:04] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:04] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:60342 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411605@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411605@127.0.0.1 [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:40:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:40:05] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:40:05] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411605@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:40:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:40:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:40:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:40:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:40:05] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:40:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:60342 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=60342 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as5b422b2e Call-ID: 1344411605@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:40:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:60342 [Aug 8 09:40:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411605@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:40:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411575@127.0.0.1' [Aug 8 09:40:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411575@127.0.0.1 [Aug 8 09:40:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411575@127.0.0.1' Method: OPTIONS [Aug 8 09:40:09] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:09] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:40:14] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:14] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:15] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:57986 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411615@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411615@127.0.0.1 [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:40:15] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:40:15] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:40:15] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411615@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:40:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:40:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:40:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:40:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:40:15] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:40:15] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:57986 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=57986 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as51906725 Call-ID: 1344411615@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:40:15] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:57986 [Aug 8 09:40:15] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411615@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:40:17] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411585@127.0.0.1' [Aug 8 09:40:17] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411585@127.0.0.1 [Aug 8 09:40:17] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411585@127.0.0.1' Method: OPTIONS [Aug 8 09:40:19] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:19] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:40:24] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:24] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:25] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:48181 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411625@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411625@127.0.0.1 [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:40:25] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:40:25] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:40:25] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411625@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:40:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:40:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:40:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:40:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:40:25] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:40:25] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:48181 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=48181 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as6649ea8d Call-ID: 1344411625@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:40:25] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:48181 [Aug 8 09:40:25] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411625@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:40:27] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411595@127.0.0.1' [Aug 8 09:40:27] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411595@127.0.0.1 [Aug 8 09:40:27] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411595@127.0.0.1' Method: OPTIONS [Aug 8 09:40:29] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:29] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 0b6ebadd11fa0b7d7f63c04073b34efa@(null) - OPTIONS (No RTP) [Aug 8 09:40:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:40:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 1210faef122d0fd37a1d5262669d15b8@127.0.0.1:5060 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6e223be4 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as5ced1d86 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 1210faef122d0fd37a1d5262669d15b8@127.0.0.1:5060 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:40:32 GMT [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:40:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6e223be4 Max-Forwards: 70 From: "Unknown" ;tag=as5ced1d86 To: Contact: Call-ID: 1210faef122d0fd37a1d5262669d15b8@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:40:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #310 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:40:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6e223be4 Max-Forwards: 70 From: "Unknown" ;tag=as5ced1d86 To: Contact: Call-ID: 1210faef122d0fd37a1d5262669d15b8@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:40:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6e223be4 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as5ced1d86 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 1210faef122d0fd37a1d5262669d15b8@127.0.0.1:5060 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:40:32 GMT [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:40:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:40:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:40:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:40:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:40:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:40:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:40:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6e223be4;received=127.0.0.1 From: "Unknown" ;tag=as5ced1d86 To: ;tag=as5ced1d86 Call-ID: 1210faef122d0fd37a1d5262669d15b8@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:40:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1210faef122d0fd37a1d5262669d15b8@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:40:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6e223be4;received=127.0.0.1 From: "Unknown" ;tag=as5ced1d86 To: ;tag=as5ced1d86 Call-ID: 1210faef122d0fd37a1d5262669d15b8@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6e223be4;received=127.0.0.1 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as5ced1d86 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as5ced1d86 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 1210faef122d0fd37a1d5262669d15b8@127.0.0.1:5060 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:40:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #310 [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '1210faef122d0fd37a1d5262669d15b8@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:40:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1210faef122d0fd37a1d5262669d15b8@127.0.0.1:5060 [Aug 8 09:40:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1210faef122d0fd37a1d5262669d15b8@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:40:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:40:34] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:34] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:35] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:53241 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411635@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411635@127.0.0.1 [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:40:35] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:40:35] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:40:35] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411635@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:40:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:40:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:40:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:40:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:40:35] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:40:35] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:53241 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=53241 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as50c55637 Call-ID: 1344411635@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:40:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:53241 [Aug 8 09:40:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411635@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:40:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411605@127.0.0.1' [Aug 8 09:40:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411605@127.0.0.1 [Aug 8 09:40:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411605@127.0.0.1' Method: OPTIONS [Aug 8 09:40:39] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:39] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:40:44] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:44] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:45] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:43076 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411645@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411645@127.0.0.1 [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:40:45] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:40:45] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:40:45] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411645@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:40:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:40:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:40:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:40:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:40:45] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:40:45] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:43076 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=43076 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as4f7e599e Call-ID: 1344411645@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:40:45] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:43076 [Aug 8 09:40:45] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411645@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:40:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411615@127.0.0.1' [Aug 8 09:40:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411615@127.0.0.1 [Aug 8 09:40:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411615@127.0.0.1' Method: OPTIONS [Aug 8 09:40:49] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:49] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:40:54] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:54] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:55] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:37827 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411655@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411655@127.0.0.1 [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:40:55] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:40:55] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:40:55] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411655@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:40:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:40:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:40:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:40:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:40:55] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:40:55] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:37827 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=37827 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as26987204 Call-ID: 1344411655@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:40:55] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:37827 [Aug 8 09:40:55] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411655@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:40:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411625@127.0.0.1' [Aug 8 09:40:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411625@127.0.0.1 [Aug 8 09:40:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411625@127.0.0.1' Method: OPTIONS [Aug 8 09:40:59] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:40:59] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:41:04] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:04] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:48581 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411665@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411665@127.0.0.1 [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:41:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:41:05] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:41:05] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411665@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:41:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:41:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:41:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:41:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:41:05] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:41:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:48581 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=48581 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as6f6925a3 Call-ID: 1344411665@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:41:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:48581 [Aug 8 09:41:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411665@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:41:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411635@127.0.0.1' [Aug 8 09:41:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411635@127.0.0.1 [Aug 8 09:41:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411635@127.0.0.1' Method: OPTIONS [Aug 8 09:41:09] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:09] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:41:14] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:14] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:15] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:37541 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411675@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411675@127.0.0.1 [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:41:15] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:41:15] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:41:15] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411675@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:41:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:41:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:41:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:41:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:41:15] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:41:15] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:37541 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=37541 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as7fa13fcd Call-ID: 1344411675@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:41:15] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:37541 [Aug 8 09:41:15] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411675@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:41:17] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411645@127.0.0.1' [Aug 8 09:41:17] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411645@127.0.0.1 [Aug 8 09:41:17] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411645@127.0.0.1' Method: OPTIONS [Aug 8 09:41:19] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:19] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:41:24] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:24] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:25] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:40019 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411685@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411685@127.0.0.1 [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:41:25] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:41:25] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:41:25] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411685@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:41:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:41:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:41:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:41:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:41:25] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:41:25] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:40019 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=40019 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as1d06ff8e Call-ID: 1344411685@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:41:25] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:40019 [Aug 8 09:41:25] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411685@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:41:27] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411655@127.0.0.1' [Aug 8 09:41:27] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411655@127.0.0.1 [Aug 8 09:41:27] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411655@127.0.0.1' Method: OPTIONS [Aug 8 09:41:29] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:29] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 6fa3664a14af53a9575ac6ca30282cd0@(null) - OPTIONS (No RTP) [Aug 8 09:41:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:41:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 431e69f5721b25b43eb2a3cc279451ba@127.0.0.1:5060 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK607ae5ab [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as72c1d336 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 431e69f5721b25b43eb2a3cc279451ba@127.0.0.1:5060 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:41:32 GMT [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:41:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK607ae5ab Max-Forwards: 70 From: "Unknown" ;tag=as72c1d336 To: Contact: Call-ID: 431e69f5721b25b43eb2a3cc279451ba@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:41:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #320 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:41:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK607ae5ab Max-Forwards: 70 From: "Unknown" ;tag=as72c1d336 To: Contact: Call-ID: 431e69f5721b25b43eb2a3cc279451ba@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:41:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK607ae5ab [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as72c1d336 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 431e69f5721b25b43eb2a3cc279451ba@127.0.0.1:5060 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:41:32 GMT [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:41:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:41:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:41:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:41:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:41:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:41:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:41:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK607ae5ab;received=127.0.0.1 From: "Unknown" ;tag=as72c1d336 To: ;tag=as72c1d336 Call-ID: 431e69f5721b25b43eb2a3cc279451ba@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:41:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '431e69f5721b25b43eb2a3cc279451ba@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:41:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK607ae5ab;received=127.0.0.1 From: "Unknown" ;tag=as72c1d336 To: ;tag=as72c1d336 Call-ID: 431e69f5721b25b43eb2a3cc279451ba@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK607ae5ab;received=127.0.0.1 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as72c1d336 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as72c1d336 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 431e69f5721b25b43eb2a3cc279451ba@127.0.0.1:5060 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:41:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #320 [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '431e69f5721b25b43eb2a3cc279451ba@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:41:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 431e69f5721b25b43eb2a3cc279451ba@127.0.0.1:5060 [Aug 8 09:41:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '431e69f5721b25b43eb2a3cc279451ba@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:41:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:41:34] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:34] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:35] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:33946 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411695@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411695@127.0.0.1 [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:41:35] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:41:35] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:41:35] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411695@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:41:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:41:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:41:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:41:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:41:35] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:41:35] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:33946 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=33946 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as1840d35e Call-ID: 1344411695@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:41:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:33946 [Aug 8 09:41:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411695@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:41:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411665@127.0.0.1' [Aug 8 09:41:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411665@127.0.0.1 [Aug 8 09:41:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411665@127.0.0.1' Method: OPTIONS [Aug 8 09:41:39] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:39] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:41:44] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:44] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:45] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:57261 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411705@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411705@127.0.0.1 [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:41:45] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:41:45] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:41:45] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411705@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:41:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:41:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:41:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:41:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:41:45] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:41:45] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:57261 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=57261 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as77e932ad Call-ID: 1344411705@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:41:45] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:57261 [Aug 8 09:41:45] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411705@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:41:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411675@127.0.0.1' [Aug 8 09:41:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411675@127.0.0.1 [Aug 8 09:41:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411675@127.0.0.1' Method: OPTIONS [Aug 8 09:41:49] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:49] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:52] DEBUG[2982] manager.c: Running action 'IAXpeers' [Aug 8 09:41:52] DEBUG[2982] manager.c: Running action 'SIPpeers' [Aug 8 09:41:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:41:54] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:54] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:55] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:55363 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411715@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411715@127.0.0.1 [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:41:55] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:41:55] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:41:55] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411715@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:41:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:41:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:41:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:41:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:41:55] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:41:55] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:55363 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=55363 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as5e42955a Call-ID: 1344411715@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:41:55] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:55363 [Aug 8 09:41:55] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411715@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:41:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411685@127.0.0.1' [Aug 8 09:41:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411685@127.0.0.1 [Aug 8 09:41:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411685@127.0.0.1' Method: OPTIONS [Aug 8 09:41:59] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:41:59] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:42:04] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:04] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:34403 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411725@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411725@127.0.0.1 [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:42:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:42:05] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:42:05] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411725@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:42:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:42:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:42:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:42:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:42:05] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:42:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:34403 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=34403 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as1de04cf5 Call-ID: 1344411725@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:42:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:34403 [Aug 8 09:42:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411725@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:42:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411695@127.0.0.1' [Aug 8 09:42:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411695@127.0.0.1 [Aug 8 09:42:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411695@127.0.0.1' Method: OPTIONS [Aug 8 09:42:09] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:09] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:42:14] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:14] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:15] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:54399 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411735@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411735@127.0.0.1 [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:42:15] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:42:15] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:42:15] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411735@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:42:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:42:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:42:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:42:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:42:15] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:42:15] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:54399 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=54399 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as37144f70 Call-ID: 1344411735@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:42:15] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:54399 [Aug 8 09:42:15] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411735@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:42:17] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411705@127.0.0.1' [Aug 8 09:42:17] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411705@127.0.0.1 [Aug 8 09:42:17] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411705@127.0.0.1' Method: OPTIONS [Aug 8 09:42:19] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:19] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:21] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> REGISTER sip:192.168.0.178 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-ru03pw80deni;rport From: "2210" ;tag=nk50x49s8u To: "2210" Call-ID: 3c3744ef3c6c-z9ngnlvl1uu6 CSeq: 2266 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.4.32 Allow-Events: dialog X-Real-IP: 192.168.2.210 Supported: path, gruu Expires: 3600 Content-Length: 0 <-------------> [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 0 [ 34]: REGISTER sip:192.168.0.178 SIP/2.0 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-ru03pw80deni;rport [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 2 [ 52]: From: "2210" ;tag=nk50x49s8u [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: "2210" [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 4 [ 34]: Call-ID: 3c3744ef3c6c-z9ngnlvl1uu6 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 5 [ 19]: CSeq: 2266 REGISTER [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 8 [ 26]: User-Agent: snom300/8.4.32 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.2.210 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 12 [ 13]: Expires: 3600 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 14 [ 0]: [Aug 8 09:42:21] VERBOSE[2371] chan_sip.c: --- (14 headers 0 lines) --- [Aug 8 09:42:21] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:42:21] DEBUG[2371] acl.c: For destination '192.168.2.210', our source address is '192.168.0.178'. [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.0.178:5060 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 3c3744ef3c6c-z9ngnlvl1uu6 - REGISTER (No RTP) [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Initializing initreq for method REGISTER - callid 3c3744ef3c6c-z9ngnlvl1uu6 [Aug 8 09:42:21] DEBUG[2371] netsock2.c: Splitting '192.168.2.210:2048' into... [Aug 8 09:42:21] DEBUG[2371] netsock2.c: ...host '192.168.2.210' and port '2048'. [Aug 8 09:42:21] VERBOSE[2371] chan_sip.c: Sending to 192.168.2.210:2048 (NAT) [Aug 8 09:42:21] DEBUG[2371] netsock2.c: Splitting '192.168.0.178' into... [Aug 8 09:42:21] DEBUG[2371] netsock2.c: ...host '192.168.0.178' and port ''. [Aug 8 09:42:21] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 192.168.2.210:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-ru03pw80deni;received=192.168.2.210;rport=2048 From: "2210" ;tag=nk50x49s8u To: "2210" ;tag=as4e1b78a7 Call-ID: 3c3744ef3c6c-z9ngnlvl1uu6 CSeq: 2266 REGISTER Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="tvox", nonce="373fddf0" Content-Length: 0 <------------> [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.2.210:2048 [Aug 8 09:42:21] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '3c3744ef3c6c-z9ngnlvl1uu6' in 32000 ms (Method: REGISTER) [Aug 8 09:42:21] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> REGISTER sip:192.168.0.178 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-atvidpusaxh6;rport From: "2210" ;tag=nk50x49s8u To: "2210" Call-ID: 3c3744ef3c6c-z9ngnlvl1uu6 CSeq: 2267 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.4.32 Allow-Events: dialog X-Real-IP: 192.168.2.210 Supported: path, gruu Authorization: Digest username="2210",realm="tvox",nonce="373fddf0",uri="sip:192.168.0.178",response="1fd0b53e6f9127f2ecbdc9ead765d845",algorithm=MD5 Expires: 3600 Content-Length: 0 <-------------> [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 0 [ 34]: REGISTER sip:192.168.0.178 SIP/2.0 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-atvidpusaxh6;rport [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 2 [ 52]: From: "2210" ;tag=nk50x49s8u [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: "2210" [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 4 [ 34]: Call-ID: 3c3744ef3c6c-z9ngnlvl1uu6 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 5 [ 19]: CSeq: 2267 REGISTER [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 8 [ 26]: User-Agent: snom300/8.4.32 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.2.210 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 12 [149]: Authorization: Digest username="2210",realm="tvox",nonce="373fddf0",uri="sip:192.168.0.178",response="1fd0b53e6f9127f2ecbdc9ead765d845",algorithm=MD5 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Header 15 [ 0]: [Aug 8 09:42:21] VERBOSE[2371] chan_sip.c: --- (15 headers 0 lines) --- [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Initializing initreq for method REGISTER - callid 3c3744ef3c6c-z9ngnlvl1uu6 [Aug 8 09:42:21] DEBUG[2371] netsock2.c: Splitting '192.168.2.210:2048' into... [Aug 8 09:42:21] DEBUG[2371] netsock2.c: ...host '192.168.2.210' and port '2048'. [Aug 8 09:42:21] VERBOSE[2371] chan_sip.c: Sending to 192.168.2.210:2048 (NAT) [Aug 8 09:42:21] DEBUG[2371] netsock2.c: Splitting '192.168.0.178' into... [Aug 8 09:42:21] DEBUG[2371] netsock2.c: ...host '192.168.0.178' and port ''. [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Aug 8 09:42:21] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-atvidpusaxh6;received=192.168.2.210;rport=2048 From: "2210" ;tag=nk50x49s8u To: "2210" ;tag=as4e1b78a7 Call-ID: 3c3744ef3c6c-z9ngnlvl1uu6 CSeq: 2267 REGISTER Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: ;expires=3600 Date: Wed, 08 Aug 2012 07:42:21 GMT Content-Length: 0 <------------> [Aug 8 09:42:21] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.2.210:2048 [Aug 8 09:42:21] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '3c3744ef3c6c-z9ngnlvl1uu6' in 32000 ms (Method: REGISTER) [Aug 8 09:42:21] DEBUG[2982] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/2210 PeerStatus: Registered Address: 192.168.2.210:2048 [Aug 8 09:42:21] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2210 [Aug 8 09:42:21] DEBUG[2321] chan_sip.c: Checking device state for peer 2210 [Aug 8 09:42:21] DEBUG[2321] devicestate.c: Changing state for SIP/2210 - state 2 (In use) [Aug 8 09:42:21] DEBUG[2321] devicestate.c: device 'SIP/2210' state '2' [Aug 8 09:42:21] DEBUG[2394] app_queue.c: Device 'SIP/2210' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 8 09:42:21] DEBUG[2982] manager.c: Running action 'SIPshowpeer' [Aug 8 09:42:21] DEBUG[2982] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:42:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:42:24] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:24] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:25] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:56772 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411745@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411745@127.0.0.1 [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:42:25] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:42:25] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:42:25] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411745@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:42:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:42:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:42:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:42:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:42:25] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:42:25] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:56772 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=56772 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as4ec6174d Call-ID: 1344411745@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:42:25] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:56772 [Aug 8 09:42:25] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411745@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:42:27] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411715@127.0.0.1' [Aug 8 09:42:27] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411715@127.0.0.1 [Aug 8 09:42:27] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411715@127.0.0.1' Method: OPTIONS [Aug 8 09:42:29] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:29] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 706dbb5e0b8622d7744c5c2e3c54fc84@(null) - OPTIONS (No RTP) [Aug 8 09:42:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:42:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 6c8e48636b901d075e9c20f162fc2323@127.0.0.1:5060 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4e861a6f [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as3e7c7ba7 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 6c8e48636b901d075e9c20f162fc2323@127.0.0.1:5060 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:42:32 GMT [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:42:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4e861a6f Max-Forwards: 70 From: "Unknown" ;tag=as3e7c7ba7 To: Contact: Call-ID: 6c8e48636b901d075e9c20f162fc2323@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:42:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #333 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:42:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4e861a6f Max-Forwards: 70 From: "Unknown" ;tag=as3e7c7ba7 To: Contact: Call-ID: 6c8e48636b901d075e9c20f162fc2323@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:42:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4e861a6f [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as3e7c7ba7 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 6c8e48636b901d075e9c20f162fc2323@127.0.0.1:5060 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:42:32 GMT [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:42:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:42:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:42:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:42:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:42:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:42:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:42:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4e861a6f;received=127.0.0.1 From: "Unknown" ;tag=as3e7c7ba7 To: ;tag=as3e7c7ba7 Call-ID: 6c8e48636b901d075e9c20f162fc2323@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:42:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '6c8e48636b901d075e9c20f162fc2323@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:42:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4e861a6f;received=127.0.0.1 From: "Unknown" ;tag=as3e7c7ba7 To: ;tag=as3e7c7ba7 Call-ID: 6c8e48636b901d075e9c20f162fc2323@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4e861a6f;received=127.0.0.1 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as3e7c7ba7 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as3e7c7ba7 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 6c8e48636b901d075e9c20f162fc2323@127.0.0.1:5060 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:42:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #333 [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '6c8e48636b901d075e9c20f162fc2323@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:42:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 6c8e48636b901d075e9c20f162fc2323@127.0.0.1:5060 [Aug 8 09:42:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '6c8e48636b901d075e9c20f162fc2323@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:42:34] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2052 ---> REGISTER sip:192.168.0.178 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-6xly2fb9bz81;rport From: "2209" ;tag=513z10vx7j To: "2209" Call-ID: 3c374457a53f-0an6y9fj4e4k CSeq: 2264 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.4.32 Allow-Events: dialog X-Real-IP: 192.168.1.102 Supported: path, gruu Expires: 3600 Content-Length: 0 <-------------> [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 0 [ 34]: REGISTER sip:192.168.0.178 SIP/2.0 [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-6xly2fb9bz81;rport [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 2 [ 52]: From: "2209" ;tag=513z10vx7j [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: "2209" [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 4 [ 34]: Call-ID: 3c374457a53f-0an6y9fj4e4k [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 5 [ 19]: CSeq: 2264 REGISTER [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 8 [ 26]: User-Agent: snom300/8.4.32 [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.102 [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 12 [ 13]: Expires: 3600 [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 14 [ 0]: [Aug 8 09:42:34] VERBOSE[2371] chan_sip.c: --- (14 headers 0 lines) --- [Aug 8 09:42:34] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:42:34] DEBUG[2371] acl.c: For destination '192.168.1.102', our source address is '192.168.0.178'. [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.0.178:5060 [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 3c374457a53f-0an6y9fj4e4k - REGISTER (No RTP) [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Initializing initreq for method REGISTER - callid 3c374457a53f-0an6y9fj4e4k [Aug 8 09:42:34] DEBUG[2371] netsock2.c: Splitting '192.168.1.102:2052' into... [Aug 8 09:42:34] DEBUG[2371] netsock2.c: ...host '192.168.1.102' and port '2052'. [Aug 8 09:42:34] VERBOSE[2371] chan_sip.c: Sending to 192.168.1.102:2052 (NAT) [Aug 8 09:42:34] DEBUG[2371] netsock2.c: Splitting '192.168.0.178' into... [Aug 8 09:42:34] DEBUG[2371] netsock2.c: ...host '192.168.0.178' and port ''. [Aug 8 09:42:34] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.102:2052 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-6xly2fb9bz81;received=192.168.1.102;rport=2052 From: "2209" ;tag=513z10vx7j To: "2209" ;tag=as372bf4e0 Call-ID: 3c374457a53f-0an6y9fj4e4k CSeq: 2264 REGISTER Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="tvox", nonce="6f8a1aec" Content-Length: 0 <------------> [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.102:2052 [Aug 8 09:42:34] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '3c374457a53f-0an6y9fj4e4k' in 32000 ms (Method: REGISTER) [Aug 8 09:42:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:42:34] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2052 ---> REGISTER sip:192.168.0.178 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-ha2xx1nemlgl;rport From: "2209" ;tag=513z10vx7j To: "2209" Call-ID: 3c374457a53f-0an6y9fj4e4k CSeq: 2265 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.4.32 Allow-Events: dialog X-Real-IP: 192.168.1.102 Supported: path, gruu Authorization: Digest username="2209",realm="tvox",nonce="6f8a1aec",uri="sip:192.168.0.178",response="07d4c86ee02af756b35f16f231ed6ed9",algorithm=MD5 Expires: 3600 Content-Length: 0 <-------------> [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 0 [ 34]: REGISTER sip:192.168.0.178 SIP/2.0 [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-ha2xx1nemlgl;rport [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 2 [ 52]: From: "2209" ;tag=513z10vx7j [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: "2209" [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 4 [ 34]: Call-ID: 3c374457a53f-0an6y9fj4e4k [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 5 [ 19]: CSeq: 2265 REGISTER [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 7 [306]: Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 8 [ 26]: User-Agent: snom300/8.4.32 [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 10 [ 24]: X-Real-IP: 192.168.1.102 [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 12 [149]: Authorization: Digest username="2209",realm="tvox",nonce="6f8a1aec",uri="sip:192.168.0.178",response="07d4c86ee02af756b35f16f231ed6ed9",algorithm=MD5 [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Header 15 [ 0]: [Aug 8 09:42:34] VERBOSE[2371] chan_sip.c: --- (15 headers 0 lines) --- [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Initializing initreq for method REGISTER - callid 3c374457a53f-0an6y9fj4e4k [Aug 8 09:42:34] DEBUG[2371] netsock2.c: Splitting '192.168.1.102:2052' into... [Aug 8 09:42:34] DEBUG[2371] netsock2.c: ...host '192.168.1.102' and port '2052'. [Aug 8 09:42:34] VERBOSE[2371] chan_sip.c: Sending to 192.168.1.102:2052 (NAT) [Aug 8 09:42:34] DEBUG[2371] netsock2.c: Splitting '192.168.0.178' into... [Aug 8 09:42:34] DEBUG[2371] netsock2.c: ...host '192.168.0.178' and port ''. [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Aug 8 09:42:34] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.102:2052 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-ha2xx1nemlgl;received=192.168.1.102;rport=2052 From: "2209" ;tag=513z10vx7j To: "2209" ;tag=as372bf4e0 Call-ID: 3c374457a53f-0an6y9fj4e4k CSeq: 2265 REGISTER Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: ;expires=3600 Date: Wed, 08 Aug 2012 07:42:34 GMT Content-Length: 0 <------------> [Aug 8 09:42:34] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.102:2052 [Aug 8 09:42:34] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '3c374457a53f-0an6y9fj4e4k' in 32000 ms (Method: REGISTER) [Aug 8 09:42:34] DEBUG[2982] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/2209 PeerStatus: Registered Address: 192.168.1.102:2052 [Aug 8 09:42:34] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2209 [Aug 8 09:42:34] DEBUG[2321] chan_sip.c: Checking device state for peer 2209 [Aug 8 09:42:34] DEBUG[2321] devicestate.c: Changing state for SIP/2209 - state 2 (In use) [Aug 8 09:42:34] DEBUG[2321] devicestate.c: device 'SIP/2209' state '2' [Aug 8 09:42:34] DEBUG[2394] app_queue.c: Device 'SIP/2209' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 8 09:42:34] DEBUG[2982] manager.c: Running action 'SIPshowpeer' [Aug 8 09:42:34] DEBUG[2982] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:42:34] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:35] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:35378 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411755@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411755@127.0.0.1 [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:42:35] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:42:35] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:42:35] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411755@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:42:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:42:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:42:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:42:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:42:35] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:42:35] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:35378 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=35378 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as3a4b0072 Call-ID: 1344411755@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:42:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:35378 [Aug 8 09:42:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411755@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:42:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411725@127.0.0.1' [Aug 8 09:42:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411725@127.0.0.1 [Aug 8 09:42:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411725@127.0.0.1' Method: OPTIONS [Aug 8 09:42:39] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:42:44] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:45] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:35309 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411765@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411765@127.0.0.1 [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:42:45] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:42:45] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:42:45] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411765@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:42:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:42:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:42:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:42:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:42:45] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:42:45] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:35309 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=35309 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as6cd33e39 Call-ID: 1344411765@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:42:45] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:35309 [Aug 8 09:42:45] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411765@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:42:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411735@127.0.0.1' [Aug 8 09:42:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411735@127.0.0.1 [Aug 8 09:42:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411735@127.0.0.1' Method: OPTIONS [Aug 8 09:42:49] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:53] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '3c3744ef3c6c-z9ngnlvl1uu6' [Aug 8 09:42:53] DEBUG[2371] chan_sip.c: Destroying SIP dialog 3c3744ef3c6c-z9ngnlvl1uu6 [Aug 8 09:42:53] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '3c3744ef3c6c-z9ngnlvl1uu6' Method: REGISTER [Aug 8 09:42:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:42:54] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:42:55] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:53335 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411775@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411775@127.0.0.1 [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:42:55] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:42:55] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:42:55] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411775@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:42:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:42:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:42:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:42:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:42:55] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:42:55] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:53335 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=53335 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as2a82c4e8 Call-ID: 1344411775@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:42:55] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:53335 [Aug 8 09:42:55] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411775@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:42:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411745@127.0.0.1' [Aug 8 09:42:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411745@127.0.0.1 [Aug 8 09:42:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411745@127.0.0.1' Method: OPTIONS [Aug 8 09:42:59] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:43:04] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:50186 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411785@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411785@127.0.0.1 [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:43:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:43:05] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:43:05] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411785@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:43:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:43:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:43:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:43:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:43:05] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:43:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:50186 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=50186 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as7bf05ca7 Call-ID: 1344411785@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:43:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:50186 [Aug 8 09:43:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411785@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:43:06] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '3c374457a53f-0an6y9fj4e4k' [Aug 8 09:43:06] DEBUG[2371] chan_sip.c: Destroying SIP dialog 3c374457a53f-0an6y9fj4e4k [Aug 8 09:43:06] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '3c374457a53f-0an6y9fj4e4k' Method: REGISTER [Aug 8 09:43:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411755@127.0.0.1' [Aug 8 09:43:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411755@127.0.0.1 [Aug 8 09:43:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411755@127.0.0.1' Method: OPTIONS [Aug 8 09:43:09] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:43:14] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:15] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:54005 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411795@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411795@127.0.0.1 [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:43:15] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:43:15] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:43:15] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411795@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:43:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:43:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:43:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:43:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:43:15] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:43:15] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:54005 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=54005 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as66f5f30e Call-ID: 1344411795@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:43:15] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:54005 [Aug 8 09:43:15] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411795@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:43:17] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411765@127.0.0.1' [Aug 8 09:43:17] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411765@127.0.0.1 [Aug 8 09:43:17] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411765@127.0.0.1' Method: OPTIONS [Aug 8 09:43:19] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:43:24] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:25] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:53194 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411805@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411805@127.0.0.1 [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:43:25] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:43:25] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:43:25] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411805@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:43:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:43:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:43:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:43:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:43:25] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:43:25] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:53194 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=53194 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as2893e379 Call-ID: 1344411805@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:43:25] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:53194 [Aug 8 09:43:25] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411805@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:43:27] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411775@127.0.0.1' [Aug 8 09:43:27] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411775@127.0.0.1 [Aug 8 09:43:27] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411775@127.0.0.1' Method: OPTIONS [Aug 8 09:43:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 080fcd9b0588d90e2bffe1905712fcce@(null) - OPTIONS (No RTP) [Aug 8 09:43:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:43:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 4198401202fbbf230eb5bd7f2023f2d4@127.0.0.1:5060 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6ba2b419 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as6fd31075 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 4198401202fbbf230eb5bd7f2023f2d4@127.0.0.1:5060 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:43:32 GMT [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:43:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6ba2b419 Max-Forwards: 70 From: "Unknown" ;tag=as6fd31075 To: Contact: Call-ID: 4198401202fbbf230eb5bd7f2023f2d4@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:43:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #346 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:43:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6ba2b419 Max-Forwards: 70 From: "Unknown" ;tag=as6fd31075 To: Contact: Call-ID: 4198401202fbbf230eb5bd7f2023f2d4@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:43:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6ba2b419 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as6fd31075 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 4198401202fbbf230eb5bd7f2023f2d4@127.0.0.1:5060 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:43:32 GMT [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:43:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:43:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:43:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:43:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:43:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:43:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:43:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6ba2b419;received=127.0.0.1 From: "Unknown" ;tag=as6fd31075 To: ;tag=as6fd31075 Call-ID: 4198401202fbbf230eb5bd7f2023f2d4@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:43:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '4198401202fbbf230eb5bd7f2023f2d4@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:43:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6ba2b419;received=127.0.0.1 From: "Unknown" ;tag=as6fd31075 To: ;tag=as6fd31075 Call-ID: 4198401202fbbf230eb5bd7f2023f2d4@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6ba2b419;received=127.0.0.1 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as6fd31075 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as6fd31075 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 4198401202fbbf230eb5bd7f2023f2d4@127.0.0.1:5060 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:43:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #346 [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '4198401202fbbf230eb5bd7f2023f2d4@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:43:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 4198401202fbbf230eb5bd7f2023f2d4@127.0.0.1:5060 [Aug 8 09:43:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '4198401202fbbf230eb5bd7f2023f2d4@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:43:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:43:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:35] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:60425 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411815@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411815@127.0.0.1 [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:43:35] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:43:35] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:43:35] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411815@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:43:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:43:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:43:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:43:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:43:35] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:43:35] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:60425 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=60425 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as5f8af9de Call-ID: 1344411815@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:43:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:60425 [Aug 8 09:43:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411815@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:43:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411785@127.0.0.1' [Aug 8 09:43:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411785@127.0.0.1 [Aug 8 09:43:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411785@127.0.0.1' Method: OPTIONS [Aug 8 09:43:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:43:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:45] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:59931 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411825@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411825@127.0.0.1 [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:43:45] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:43:45] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:43:45] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411825@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:43:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:43:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:43:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:43:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:43:45] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:43:45] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:59931 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=59931 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as003158b6 Call-ID: 1344411825@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:43:45] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:59931 [Aug 8 09:43:45] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411825@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:43:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411795@127.0.0.1' [Aug 8 09:43:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411795@127.0.0.1 [Aug 8 09:43:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411795@127.0.0.1' Method: OPTIONS [Aug 8 09:43:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:43:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:43:55] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:35458 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411835@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411835@127.0.0.1 [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:43:55] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:43:55] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:43:55] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411835@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:43:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:43:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:43:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:43:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:43:55] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:43:55] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:35458 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=35458 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as066e1c0f Call-ID: 1344411835@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:43:55] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:35458 [Aug 8 09:43:55] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411835@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:43:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411805@127.0.0.1' [Aug 8 09:43:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411805@127.0.0.1 [Aug 8 09:43:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411805@127.0.0.1' Method: OPTIONS [Aug 8 09:44:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:44:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:59592 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411845@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411845@127.0.0.1 [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:44:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:44:05] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:44:05] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411845@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:44:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:44:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:44:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:44:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:44:05] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:44:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:59592 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=59592 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as41f7542f Call-ID: 1344411845@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:44:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:59592 [Aug 8 09:44:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411845@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:44:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411815@127.0.0.1' [Aug 8 09:44:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411815@127.0.0.1 [Aug 8 09:44:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411815@127.0.0.1' Method: OPTIONS [Aug 8 09:44:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:44:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:15] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:39930 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411855@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411855@127.0.0.1 [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:44:15] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:44:15] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:44:15] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411855@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:44:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:44:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:44:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:44:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:44:15] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:44:15] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:39930 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=39930 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as4ea426bc Call-ID: 1344411855@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:44:15] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:39930 [Aug 8 09:44:15] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411855@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:44:17] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411825@127.0.0.1' [Aug 8 09:44:17] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411825@127.0.0.1 [Aug 8 09:44:17] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411825@127.0.0.1' Method: OPTIONS [Aug 8 09:44:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:44:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:25] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:50475 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411865@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411865@127.0.0.1 [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:44:25] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:44:25] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:44:25] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411865@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:44:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:44:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:44:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:44:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:44:25] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:44:25] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:50475 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=50475 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as5be44ad9 Call-ID: 1344411865@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:44:25] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:50475 [Aug 8 09:44:25] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411865@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:44:27] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411835@127.0.0.1' [Aug 8 09:44:27] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411835@127.0.0.1 [Aug 8 09:44:27] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411835@127.0.0.1' Method: OPTIONS [Aug 8 09:44:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 5022c31f2e37f17903e498725feb39c2@(null) - OPTIONS (No RTP) [Aug 8 09:44:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:44:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 1ee2e7345481b17f17e3d152540edeae@127.0.0.1:5060 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5dd357f7 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as3d6f7ed7 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 1ee2e7345481b17f17e3d152540edeae@127.0.0.1:5060 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:44:32 GMT [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:44:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5dd357f7 Max-Forwards: 70 From: "Unknown" ;tag=as3d6f7ed7 To: Contact: Call-ID: 1ee2e7345481b17f17e3d152540edeae@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:44:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #356 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:44:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5dd357f7 Max-Forwards: 70 From: "Unknown" ;tag=as3d6f7ed7 To: Contact: Call-ID: 1ee2e7345481b17f17e3d152540edeae@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:44:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5dd357f7 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as3d6f7ed7 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 1ee2e7345481b17f17e3d152540edeae@127.0.0.1:5060 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:44:32 GMT [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:44:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:44:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:44:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:44:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:44:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:44:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:44:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5dd357f7;received=127.0.0.1 From: "Unknown" ;tag=as3d6f7ed7 To: ;tag=as3d6f7ed7 Call-ID: 1ee2e7345481b17f17e3d152540edeae@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:44:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1ee2e7345481b17f17e3d152540edeae@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:44:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5dd357f7;received=127.0.0.1 From: "Unknown" ;tag=as3d6f7ed7 To: ;tag=as3d6f7ed7 Call-ID: 1ee2e7345481b17f17e3d152540edeae@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5dd357f7;received=127.0.0.1 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as3d6f7ed7 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as3d6f7ed7 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 1ee2e7345481b17f17e3d152540edeae@127.0.0.1:5060 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:44:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #356 [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '1ee2e7345481b17f17e3d152540edeae@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:44:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1ee2e7345481b17f17e3d152540edeae@127.0.0.1:5060 [Aug 8 09:44:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1ee2e7345481b17f17e3d152540edeae@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:44:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:44:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:35] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:41723 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411875@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411875@127.0.0.1 [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:44:35] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:44:35] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:44:35] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411875@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:44:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:44:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:44:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:44:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:44:35] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:44:35] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:41723 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=41723 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as5334ef6b Call-ID: 1344411875@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:44:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:41723 [Aug 8 09:44:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411875@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:44:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411845@127.0.0.1' [Aug 8 09:44:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411845@127.0.0.1 [Aug 8 09:44:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411845@127.0.0.1' Method: OPTIONS [Aug 8 09:44:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:44:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:45] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:55172 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411885@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411885@127.0.0.1 [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:44:45] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:44:45] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:44:45] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411885@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:44:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:44:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:44:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:44:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:44:45] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:44:45] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:55172 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=55172 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as27f4ac67 Call-ID: 1344411885@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:44:45] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:55172 [Aug 8 09:44:45] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411885@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:44:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411855@127.0.0.1' [Aug 8 09:44:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411855@127.0.0.1 [Aug 8 09:44:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411855@127.0.0.1' Method: OPTIONS [Aug 8 09:44:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:44:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:44:55] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:40287 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411895@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411895@127.0.0.1 [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:44:55] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:44:55] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:44:55] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411895@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:44:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:44:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:44:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:44:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:44:55] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:44:55] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:40287 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=40287 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as0bb0c951 Call-ID: 1344411895@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:44:55] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:40287 [Aug 8 09:44:55] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411895@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:44:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411865@127.0.0.1' [Aug 8 09:44:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411865@127.0.0.1 [Aug 8 09:44:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411865@127.0.0.1' Method: OPTIONS [Aug 8 09:45:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:45:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:56039 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411905@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411905@127.0.0.1 [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:45:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:45:05] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:45:05] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411905@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:45:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:45:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:45:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:45:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:45:05] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:45:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:56039 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=56039 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as358c2dc8 Call-ID: 1344411905@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:45:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:56039 [Aug 8 09:45:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411905@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:45:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411875@127.0.0.1' [Aug 8 09:45:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411875@127.0.0.1 [Aug 8 09:45:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411875@127.0.0.1' Method: OPTIONS [Aug 8 09:45:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:45:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:15] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:51089 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411915@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411915@127.0.0.1 [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:45:15] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:45:15] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:45:15] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411915@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:45:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:45:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:45:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:45:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:45:15] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:45:15] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:51089 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=51089 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as1d885038 Call-ID: 1344411915@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:45:15] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:51089 [Aug 8 09:45:15] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411915@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:45:17] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411885@127.0.0.1' [Aug 8 09:45:17] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411885@127.0.0.1 [Aug 8 09:45:17] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411885@127.0.0.1' Method: OPTIONS [Aug 8 09:45:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:45:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:25] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:33598 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411925@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411925@127.0.0.1 [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:45:25] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:45:25] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:45:25] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411925@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:45:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:45:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:45:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:45:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:45:25] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:45:25] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:33598 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=33598 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as458794e0 Call-ID: 1344411925@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:45:25] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:33598 [Aug 8 09:45:25] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411925@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:45:27] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411895@127.0.0.1' [Aug 8 09:45:27] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411895@127.0.0.1 [Aug 8 09:45:27] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411895@127.0.0.1' Method: OPTIONS [Aug 8 09:45:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 313e2c7e25b356a458a64d9d369e4f91@(null) - OPTIONS (No RTP) [Aug 8 09:45:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:45:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 70327cfc5a709ed80f162b1574c0c719@127.0.0.1:5060 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK1987e954 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as6915bd5f [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 70327cfc5a709ed80f162b1574c0c719@127.0.0.1:5060 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:45:32 GMT [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:45:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK1987e954 Max-Forwards: 70 From: "Unknown" ;tag=as6915bd5f To: Contact: Call-ID: 70327cfc5a709ed80f162b1574c0c719@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:45:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #366 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:45:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK1987e954 Max-Forwards: 70 From: "Unknown" ;tag=as6915bd5f To: Contact: Call-ID: 70327cfc5a709ed80f162b1574c0c719@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:45:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK1987e954 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as6915bd5f [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 70327cfc5a709ed80f162b1574c0c719@127.0.0.1:5060 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:45:32 GMT [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:45:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:45:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:45:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:45:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:45:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:45:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:45:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK1987e954;received=127.0.0.1 From: "Unknown" ;tag=as6915bd5f To: ;tag=as6915bd5f Call-ID: 70327cfc5a709ed80f162b1574c0c719@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:45:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '70327cfc5a709ed80f162b1574c0c719@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:45:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK1987e954;received=127.0.0.1 From: "Unknown" ;tag=as6915bd5f To: ;tag=as6915bd5f Call-ID: 70327cfc5a709ed80f162b1574c0c719@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK1987e954;received=127.0.0.1 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as6915bd5f [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as6915bd5f [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 70327cfc5a709ed80f162b1574c0c719@127.0.0.1:5060 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:45:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #366 [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '70327cfc5a709ed80f162b1574c0c719@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:45:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 70327cfc5a709ed80f162b1574c0c719@127.0.0.1:5060 [Aug 8 09:45:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '70327cfc5a709ed80f162b1574c0c719@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:45:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:45:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:35] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:37446 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411935@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411935@127.0.0.1 [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:45:35] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:45:35] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:45:35] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411935@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:45:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:45:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:45:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:45:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:45:35] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:45:35] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:37446 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=37446 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as1ca157f3 Call-ID: 1344411935@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:45:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:37446 [Aug 8 09:45:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411935@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:45:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411905@127.0.0.1' [Aug 8 09:45:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411905@127.0.0.1 [Aug 8 09:45:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411905@127.0.0.1' Method: OPTIONS [Aug 8 09:45:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:45:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:45] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:41824 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411945@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411945@127.0.0.1 [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:45:45] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:45:45] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:45:45] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411945@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:45:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:45:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:45:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:45:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:45:45] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:45:45] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:41824 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=41824 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as51bc0a96 Call-ID: 1344411945@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:45:45] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:41824 [Aug 8 09:45:45] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411945@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:45:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411915@127.0.0.1' [Aug 8 09:45:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411915@127.0.0.1 [Aug 8 09:45:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411915@127.0.0.1' Method: OPTIONS [Aug 8 09:45:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:45:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:45:55] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:36738 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411955@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411955@127.0.0.1 [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:45:55] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:45:55] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:45:55] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411955@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:45:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:45:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:45:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:45:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:45:55] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:45:55] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:36738 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=36738 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as233a146a Call-ID: 1344411955@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:45:55] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:36738 [Aug 8 09:45:55] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411955@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:45:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411925@127.0.0.1' [Aug 8 09:45:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411925@127.0.0.1 [Aug 8 09:45:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411925@127.0.0.1' Method: OPTIONS [Aug 8 09:46:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:46:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:49663 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411965@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411965@127.0.0.1 [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:46:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:46:05] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:46:05] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411965@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:46:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:46:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:46:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:46:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:46:05] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:46:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:49663 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=49663 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as2863b314 Call-ID: 1344411965@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:46:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:49663 [Aug 8 09:46:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411965@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:46:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411935@127.0.0.1' [Aug 8 09:46:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411935@127.0.0.1 [Aug 8 09:46:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411935@127.0.0.1' Method: OPTIONS [Aug 8 09:46:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:46:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:15] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:33883 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411975@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411975@127.0.0.1 [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:46:15] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:46:15] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:46:15] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411975@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:46:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:46:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:46:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:46:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:46:15] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:46:15] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:33883 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=33883 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as74644a0c Call-ID: 1344411975@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:46:15] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:33883 [Aug 8 09:46:15] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411975@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:46:17] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411945@127.0.0.1' [Aug 8 09:46:17] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411945@127.0.0.1 [Aug 8 09:46:17] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411945@127.0.0.1' Method: OPTIONS [Aug 8 09:46:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:46:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:25] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:51291 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411985@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411985@127.0.0.1 [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:46:25] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:46:25] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:46:25] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411985@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:46:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:46:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:46:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:46:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:46:25] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:46:25] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:51291 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=51291 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as3507d709 Call-ID: 1344411985@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:46:25] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:51291 [Aug 8 09:46:25] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411985@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:46:27] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411955@127.0.0.1' [Aug 8 09:46:27] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411955@127.0.0.1 [Aug 8 09:46:27] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411955@127.0.0.1' Method: OPTIONS [Aug 8 09:46:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 289ebadf6821b9fe2ab3080b1d692b55@(null) - OPTIONS (No RTP) [Aug 8 09:46:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:46:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 23a0bf6879c77ae03b8b2719093f8fd2@127.0.0.1:5060 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK212f07bc [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as0d27a388 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 23a0bf6879c77ae03b8b2719093f8fd2@127.0.0.1:5060 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:46:32 GMT [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:46:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK212f07bc Max-Forwards: 70 From: "Unknown" ;tag=as0d27a388 To: Contact: Call-ID: 23a0bf6879c77ae03b8b2719093f8fd2@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:46:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #376 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:46:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK212f07bc Max-Forwards: 70 From: "Unknown" ;tag=as0d27a388 To: Contact: Call-ID: 23a0bf6879c77ae03b8b2719093f8fd2@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:46:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK212f07bc [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as0d27a388 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 23a0bf6879c77ae03b8b2719093f8fd2@127.0.0.1:5060 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:46:32 GMT [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:46:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:46:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:46:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:46:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:46:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:46:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:46:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK212f07bc;received=127.0.0.1 From: "Unknown" ;tag=as0d27a388 To: ;tag=as0d27a388 Call-ID: 23a0bf6879c77ae03b8b2719093f8fd2@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:46:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '23a0bf6879c77ae03b8b2719093f8fd2@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:46:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK212f07bc;received=127.0.0.1 From: "Unknown" ;tag=as0d27a388 To: ;tag=as0d27a388 Call-ID: 23a0bf6879c77ae03b8b2719093f8fd2@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK212f07bc;received=127.0.0.1 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as0d27a388 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as0d27a388 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 23a0bf6879c77ae03b8b2719093f8fd2@127.0.0.1:5060 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:46:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #376 [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '23a0bf6879c77ae03b8b2719093f8fd2@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:46:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 23a0bf6879c77ae03b8b2719093f8fd2@127.0.0.1:5060 [Aug 8 09:46:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '23a0bf6879c77ae03b8b2719093f8fd2@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:46:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:46:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:35] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:52124 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344411995@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344411995@127.0.0.1 [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:46:35] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:46:35] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:46:35] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344411995@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:46:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:46:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:46:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:46:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:46:35] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:46:35] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:52124 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=52124 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as3fd62121 Call-ID: 1344411995@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:46:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:52124 [Aug 8 09:46:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344411995@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:46:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411965@127.0.0.1' [Aug 8 09:46:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411965@127.0.0.1 [Aug 8 09:46:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411965@127.0.0.1' Method: OPTIONS [Aug 8 09:46:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:46:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:45] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:54098 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412005@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412005@127.0.0.1 [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:46:45] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:46:45] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:46:45] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412005@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:46:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:46:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:46:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:46:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:46:45] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:46:45] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:54098 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=54098 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as4185f0f8 Call-ID: 1344412005@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:46:45] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:54098 [Aug 8 09:46:45] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412005@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:46:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411975@127.0.0.1' [Aug 8 09:46:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411975@127.0.0.1 [Aug 8 09:46:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411975@127.0.0.1' Method: OPTIONS [Aug 8 09:46:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:52] DEBUG[2982] manager.c: Running action 'IAXpeers' [Aug 8 09:46:52] DEBUG[2982] manager.c: Running action 'SIPpeers' [Aug 8 09:46:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:46:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:46:55] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:46244 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412015@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412015@127.0.0.1 [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:46:55] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:46:55] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:46:55] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412015@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:46:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:46:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:46:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:46:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:46:55] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:46:55] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:46244 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=46244 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as491fa569 Call-ID: 1344412015@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:46:55] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:46244 [Aug 8 09:46:55] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412015@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:46:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411985@127.0.0.1' [Aug 8 09:46:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411985@127.0.0.1 [Aug 8 09:46:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411985@127.0.0.1' Method: OPTIONS [Aug 8 09:47:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:47:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:49465 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412025@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412025@127.0.0.1 [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:47:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:47:05] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:47:05] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412025@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:47:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:47:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:47:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:47:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:47:05] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:47:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:49465 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=49465 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as386cee01 Call-ID: 1344412025@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:47:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:49465 [Aug 8 09:47:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412025@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:47:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344411995@127.0.0.1' [Aug 8 09:47:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344411995@127.0.0.1 [Aug 8 09:47:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344411995@127.0.0.1' Method: OPTIONS [Aug 8 09:47:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:47:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:15] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:50787 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412035@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412035@127.0.0.1 [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:47:15] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:47:15] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:47:15] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412035@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:47:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:47:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:47:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:47:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:47:15] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:47:15] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:50787 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=50787 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as1079ac59 Call-ID: 1344412035@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:47:15] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:50787 [Aug 8 09:47:15] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412035@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:47:17] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412005@127.0.0.1' [Aug 8 09:47:17] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412005@127.0.0.1 [Aug 8 09:47:17] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412005@127.0.0.1' Method: OPTIONS [Aug 8 09:47:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:47:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:25] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:52736 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412045@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412045@127.0.0.1 [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:47:25] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:47:25] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:47:25] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412045@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:47:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:47:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:47:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:47:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:47:25] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:47:25] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:52736 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=52736 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as52da34af Call-ID: 1344412045@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:47:25] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:52736 [Aug 8 09:47:25] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412045@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:47:27] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412015@127.0.0.1' [Aug 8 09:47:27] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412015@127.0.0.1 [Aug 8 09:47:27] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412015@127.0.0.1' Method: OPTIONS [Aug 8 09:47:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1699c1f75e0376f529297a681349d89f@(null) - OPTIONS (No RTP) [Aug 8 09:47:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:47:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 4349527758dfcba843eed9d01f53cb90@127.0.0.1:5060 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4b022f09 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as5778d72e [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 4349527758dfcba843eed9d01f53cb90@127.0.0.1:5060 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:47:32 GMT [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:47:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4b022f09 Max-Forwards: 70 From: "Unknown" ;tag=as5778d72e To: Contact: Call-ID: 4349527758dfcba843eed9d01f53cb90@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:47:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #386 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:47:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4b022f09 Max-Forwards: 70 From: "Unknown" ;tag=as5778d72e To: Contact: Call-ID: 4349527758dfcba843eed9d01f53cb90@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:47:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4b022f09 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as5778d72e [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 4349527758dfcba843eed9d01f53cb90@127.0.0.1:5060 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:47:32 GMT [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:47:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:47:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:47:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:47:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:47:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:47:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:47:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4b022f09;received=127.0.0.1 From: "Unknown" ;tag=as5778d72e To: ;tag=as5778d72e Call-ID: 4349527758dfcba843eed9d01f53cb90@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:47:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '4349527758dfcba843eed9d01f53cb90@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:47:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4b022f09;received=127.0.0.1 From: "Unknown" ;tag=as5778d72e To: ;tag=as5778d72e Call-ID: 4349527758dfcba843eed9d01f53cb90@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4b022f09;received=127.0.0.1 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as5778d72e [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as5778d72e [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 4349527758dfcba843eed9d01f53cb90@127.0.0.1:5060 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:47:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #386 [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '4349527758dfcba843eed9d01f53cb90@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:47:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 4349527758dfcba843eed9d01f53cb90@127.0.0.1:5060 [Aug 8 09:47:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '4349527758dfcba843eed9d01f53cb90@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:47:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:47:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:35] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:58763 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412055@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412055@127.0.0.1 [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:47:35] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:47:35] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:47:35] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412055@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:47:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:47:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:47:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:47:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:47:35] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:47:35] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:58763 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=58763 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as05b3171d Call-ID: 1344412055@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:47:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:58763 [Aug 8 09:47:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412055@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:47:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412025@127.0.0.1' [Aug 8 09:47:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412025@127.0.0.1 [Aug 8 09:47:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412025@127.0.0.1' Method: OPTIONS [Aug 8 09:47:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:47:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:45] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:56818 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412065@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412065@127.0.0.1 [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:47:45] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:47:45] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:47:45] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412065@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:47:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:47:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:47:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:47:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:47:45] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:47:45] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:56818 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=56818 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as22e45928 Call-ID: 1344412065@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:47:45] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:56818 [Aug 8 09:47:45] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412065@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:47:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412035@127.0.0.1' [Aug 8 09:47:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412035@127.0.0.1 [Aug 8 09:47:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412035@127.0.0.1' Method: OPTIONS [Aug 8 09:47:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:47:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:47:55] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:35995 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412075@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412075@127.0.0.1 [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:47:55] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:47:55] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:47:55] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412075@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:47:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:47:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:47:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:47:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:47:55] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:47:55] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:35995 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=35995 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as13b5f5fa Call-ID: 1344412075@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:47:55] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:35995 [Aug 8 09:47:55] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412075@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:47:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412045@127.0.0.1' [Aug 8 09:47:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412045@127.0.0.1 [Aug 8 09:47:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412045@127.0.0.1' Method: OPTIONS [Aug 8 09:48:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:48:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:40913 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412085@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412085@127.0.0.1 [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:48:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:48:05] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:48:05] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412085@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:48:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:48:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:48:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:48:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:48:05] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:48:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:40913 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=40913 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as02d1d8a0 Call-ID: 1344412085@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:48:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:40913 [Aug 8 09:48:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412085@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:48:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412055@127.0.0.1' [Aug 8 09:48:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412055@127.0.0.1 [Aug 8 09:48:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412055@127.0.0.1' Method: OPTIONS [Aug 8 09:48:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:48:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:15] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:41730 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412095@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412095@127.0.0.1 [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:48:15] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:48:15] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:48:15] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412095@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:48:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:48:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:48:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:48:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:48:15] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:48:15] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:41730 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=41730 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as43f633c7 Call-ID: 1344412095@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:48:15] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:41730 [Aug 8 09:48:15] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412095@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:48:17] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412065@127.0.0.1' [Aug 8 09:48:17] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412065@127.0.0.1 [Aug 8 09:48:17] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412065@127.0.0.1' Method: OPTIONS [Aug 8 09:48:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:48:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:25] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:57701 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412105@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412105@127.0.0.1 [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:48:25] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:48:25] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:48:25] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412105@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:48:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:48:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:48:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:48:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:48:25] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:48:25] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:57701 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=57701 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as20efa7e2 Call-ID: 1344412105@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:48:25] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:57701 [Aug 8 09:48:25] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412105@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:48:27] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412075@127.0.0.1' [Aug 8 09:48:27] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412075@127.0.0.1 [Aug 8 09:48:27] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412075@127.0.0.1' Method: OPTIONS [Aug 8 09:48:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 001718485a42a1a12c43ad85213006ec@(null) - OPTIONS (No RTP) [Aug 8 09:48:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:48:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 617b0d1d68c4ecec31d454f213a0cbee@127.0.0.1:5060 [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0852859a [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as35e8244f [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 617b0d1d68c4ecec31d454f213a0cbee@127.0.0.1:5060 [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:48:32 GMT [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:48:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0852859a Max-Forwards: 70 From: "Unknown" ;tag=as35e8244f To: Contact: Call-ID: 617b0d1d68c4ecec31d454f213a0cbee@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:48:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #396 [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:48:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0852859a Max-Forwards: 70 From: "Unknown" ;tag=as35e8244f To: Contact: Call-ID: 617b0d1d68c4ecec31d454f213a0cbee@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:48:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0852859a [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as35e8244f [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 617b0d1d68c4ecec31d454f213a0cbee@127.0.0.1:5060 [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:48:32 GMT [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:48:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:48:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:48:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:48:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:48:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:48:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:48:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0852859a;received=127.0.0.1 From: "Unknown" ;tag=as35e8244f To: ;tag=as35e8244f Call-ID: 617b0d1d68c4ecec31d454f213a0cbee@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:48:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '617b0d1d68c4ecec31d454f213a0cbee@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:48:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0852859a;received=127.0.0.1 From: "Unknown" ;tag=as35e8244f To: ;tag=as35e8244f Call-ID: 617b0d1d68c4ecec31d454f213a0cbee@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0852859a;received=127.0.0.1 [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as35e8244f [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as35e8244f [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 617b0d1d68c4ecec31d454f213a0cbee@127.0.0.1:5060 [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:48:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #396 [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '617b0d1d68c4ecec31d454f213a0cbee@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:48:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 617b0d1d68c4ecec31d454f213a0cbee@127.0.0.1:5060 [Aug 8 09:48:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '617b0d1d68c4ecec31d454f213a0cbee@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:48:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:48:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:35] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:33669 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412115@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412115@127.0.0.1 [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:48:35] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:48:35] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:48:35] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412115@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:48:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:48:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:48:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:48:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:48:35] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:48:35] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:33669 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=33669 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as5c1db578 Call-ID: 1344412115@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:48:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:33669 [Aug 8 09:48:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412115@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:48:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412085@127.0.0.1' [Aug 8 09:48:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412085@127.0.0.1 [Aug 8 09:48:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412085@127.0.0.1' Method: OPTIONS [Aug 8 09:48:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:48:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:45] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:54456 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412125@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412125@127.0.0.1 [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:48:45] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:48:45] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:48:45] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412125@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:48:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:48:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:48:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:48:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:48:45] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:48:45] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:54456 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=54456 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as6bb39c4a Call-ID: 1344412125@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:48:45] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:54456 [Aug 8 09:48:45] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412125@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:48:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412095@127.0.0.1' [Aug 8 09:48:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412095@127.0.0.1 [Aug 8 09:48:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412095@127.0.0.1' Method: OPTIONS [Aug 8 09:48:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:48:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:48:55] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:50034 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412135@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412135@127.0.0.1 [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:48:55] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:48:55] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:48:55] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412135@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:48:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:48:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:48:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:48:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:48:55] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:48:55] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:50034 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=50034 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as32476504 Call-ID: 1344412135@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:48:55] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:50034 [Aug 8 09:48:55] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412135@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:48:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412105@127.0.0.1' [Aug 8 09:48:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412105@127.0.0.1 [Aug 8 09:48:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412105@127.0.0.1' Method: OPTIONS [Aug 8 09:49:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:49:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:33194 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412145@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412145@127.0.0.1 [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:49:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:49:05] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:49:05] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412145@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:49:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:49:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:49:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:49:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:49:05] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:49:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:33194 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=33194 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as106cd322 Call-ID: 1344412145@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:49:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:33194 [Aug 8 09:49:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412145@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:49:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412115@127.0.0.1' [Aug 8 09:49:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412115@127.0.0.1 [Aug 8 09:49:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412115@127.0.0.1' Method: OPTIONS [Aug 8 09:49:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:49:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:15] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:48158 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412155@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412155@127.0.0.1 [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:49:15] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:49:15] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:49:15] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412155@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:49:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:49:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:49:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:49:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:49:15] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:49:15] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:48158 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=48158 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as41b03db3 Call-ID: 1344412155@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:49:15] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:48158 [Aug 8 09:49:15] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412155@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:49:17] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412125@127.0.0.1' [Aug 8 09:49:17] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412125@127.0.0.1 [Aug 8 09:49:17] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412125@127.0.0.1' Method: OPTIONS [Aug 8 09:49:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:49:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:25] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:38437 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412165@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412165@127.0.0.1 [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:49:25] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:49:25] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:49:25] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412165@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:49:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:49:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:49:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:49:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:49:25] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:49:25] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:38437 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=38437 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as2bdc44d8 Call-ID: 1344412165@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:49:25] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:38437 [Aug 8 09:49:25] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412165@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:49:27] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412135@127.0.0.1' [Aug 8 09:49:27] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412135@127.0.0.1 [Aug 8 09:49:27] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412135@127.0.0.1' Method: OPTIONS [Aug 8 09:49:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 2ed04a104e3d61fe269cc4527ba08096@(null) - OPTIONS (No RTP) [Aug 8 09:49:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:49:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 6be13e875225bef63a034b0018064de3@127.0.0.1:5060 [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5a354e2d [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as34dbdfee [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 6be13e875225bef63a034b0018064de3@127.0.0.1:5060 [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:49:32 GMT [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:49:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5a354e2d Max-Forwards: 70 From: "Unknown" ;tag=as34dbdfee To: Contact: Call-ID: 6be13e875225bef63a034b0018064de3@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:49:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #406 [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:49:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5a354e2d Max-Forwards: 70 From: "Unknown" ;tag=as34dbdfee To: Contact: Call-ID: 6be13e875225bef63a034b0018064de3@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:49:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5a354e2d [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as34dbdfee [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 6be13e875225bef63a034b0018064de3@127.0.0.1:5060 [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:49:32 GMT [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:49:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:49:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:49:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:49:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:49:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:49:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:49:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5a354e2d;received=127.0.0.1 From: "Unknown" ;tag=as34dbdfee To: ;tag=as34dbdfee Call-ID: 6be13e875225bef63a034b0018064de3@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:49:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '6be13e875225bef63a034b0018064de3@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:49:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5a354e2d;received=127.0.0.1 From: "Unknown" ;tag=as34dbdfee To: ;tag=as34dbdfee Call-ID: 6be13e875225bef63a034b0018064de3@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5a354e2d;received=127.0.0.1 [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as34dbdfee [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as34dbdfee [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 6be13e875225bef63a034b0018064de3@127.0.0.1:5060 [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:49:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #406 [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '6be13e875225bef63a034b0018064de3@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:49:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 6be13e875225bef63a034b0018064de3@127.0.0.1:5060 [Aug 8 09:49:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '6be13e875225bef63a034b0018064de3@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:49:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:49:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:35] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:60788 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412175@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412175@127.0.0.1 [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:49:35] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:49:35] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:49:35] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412175@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:49:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:49:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:49:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:49:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:49:35] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:49:35] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:60788 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=60788 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as4cc60480 Call-ID: 1344412175@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:49:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:60788 [Aug 8 09:49:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412175@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:49:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412145@127.0.0.1' [Aug 8 09:49:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412145@127.0.0.1 [Aug 8 09:49:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412145@127.0.0.1' Method: OPTIONS [Aug 8 09:49:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:49:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:45] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:50802 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412185@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412185@127.0.0.1 [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:49:45] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:49:45] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:49:45] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412185@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:49:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:49:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:49:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:49:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:49:45] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:49:45] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:50802 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=50802 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as2e315883 Call-ID: 1344412185@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:49:45] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:50802 [Aug 8 09:49:45] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412185@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:49:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412155@127.0.0.1' [Aug 8 09:49:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412155@127.0.0.1 [Aug 8 09:49:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412155@127.0.0.1' Method: OPTIONS [Aug 8 09:49:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:49:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:49:55] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:41850 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412195@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412195@127.0.0.1 [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:49:55] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:49:55] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:49:55] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412195@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:49:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:49:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:49:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:49:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:49:55] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:49:55] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:41850 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=41850 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as4b697d19 Call-ID: 1344412195@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:49:55] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:41850 [Aug 8 09:49:55] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412195@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:49:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412165@127.0.0.1' [Aug 8 09:49:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412165@127.0.0.1 [Aug 8 09:49:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412165@127.0.0.1' Method: OPTIONS [Aug 8 09:50:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:50:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:37329 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412205@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412205@127.0.0.1 [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:50:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:50:05] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:50:05] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412205@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:50:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:50:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:50:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:50:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:50:05] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:50:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:37329 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=37329 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as42b97fa8 Call-ID: 1344412205@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:50:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:37329 [Aug 8 09:50:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412205@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:50:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412175@127.0.0.1' [Aug 8 09:50:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412175@127.0.0.1 [Aug 8 09:50:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412175@127.0.0.1' Method: OPTIONS [Aug 8 09:50:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:50:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:15] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:58221 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412215@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412215@127.0.0.1 [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:50:15] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:50:15] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:50:15] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412215@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:50:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:50:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:50:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:50:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:50:15] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:50:15] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:58221 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=58221 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as4fe2b80d Call-ID: 1344412215@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:50:15] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:58221 [Aug 8 09:50:15] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412215@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:50:17] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412185@127.0.0.1' [Aug 8 09:50:17] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412185@127.0.0.1 [Aug 8 09:50:17] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412185@127.0.0.1' Method: OPTIONS [Aug 8 09:50:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:50:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:25] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:53961 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412225@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412225@127.0.0.1 [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:50:25] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:50:25] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:50:25] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412225@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:50:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:50:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:50:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:50:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:50:25] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:50:25] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:53961 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=53961 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as787a3d7b Call-ID: 1344412225@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:50:25] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:53961 [Aug 8 09:50:25] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412225@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:50:27] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412195@127.0.0.1' [Aug 8 09:50:27] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412195@127.0.0.1 [Aug 8 09:50:27] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412195@127.0.0.1' Method: OPTIONS [Aug 8 09:50:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 6c35f4da34f8591e44e35dd05ab40481@(null) - OPTIONS (No RTP) [Aug 8 09:50:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:50:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 051149a0527773de286d096203c3ecf2@127.0.0.1:5060 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4f50fdbb [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as4fefc6b0 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 051149a0527773de286d096203c3ecf2@127.0.0.1:5060 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:50:32 GMT [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:50:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4f50fdbb Max-Forwards: 70 From: "Unknown" ;tag=as4fefc6b0 To: Contact: Call-ID: 051149a0527773de286d096203c3ecf2@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:50:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #416 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:50:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4f50fdbb Max-Forwards: 70 From: "Unknown" ;tag=as4fefc6b0 To: Contact: Call-ID: 051149a0527773de286d096203c3ecf2@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:50:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4f50fdbb [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as4fefc6b0 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 051149a0527773de286d096203c3ecf2@127.0.0.1:5060 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:50:32 GMT [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:50:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:50:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:50:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:50:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:50:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:50:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:50:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4f50fdbb;received=127.0.0.1 From: "Unknown" ;tag=as4fefc6b0 To: ;tag=as4fefc6b0 Call-ID: 051149a0527773de286d096203c3ecf2@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:50:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '051149a0527773de286d096203c3ecf2@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:50:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4f50fdbb;received=127.0.0.1 From: "Unknown" ;tag=as4fefc6b0 To: ;tag=as4fefc6b0 Call-ID: 051149a0527773de286d096203c3ecf2@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4f50fdbb;received=127.0.0.1 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as4fefc6b0 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as4fefc6b0 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 051149a0527773de286d096203c3ecf2@127.0.0.1:5060 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:50:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #416 [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '051149a0527773de286d096203c3ecf2@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:50:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 051149a0527773de286d096203c3ecf2@127.0.0.1:5060 [Aug 8 09:50:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '051149a0527773de286d096203c3ecf2@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:50:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:50:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:35] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:39630 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412235@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412235@127.0.0.1 [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:50:35] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:50:35] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:50:35] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412235@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:50:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:50:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:50:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:50:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:50:35] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:50:35] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:39630 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=39630 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as20e9422a Call-ID: 1344412235@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:50:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:39630 [Aug 8 09:50:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412235@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:50:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412205@127.0.0.1' [Aug 8 09:50:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412205@127.0.0.1 [Aug 8 09:50:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412205@127.0.0.1' Method: OPTIONS [Aug 8 09:50:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:50:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:45] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:38499 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412245@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412245@127.0.0.1 [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:50:45] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:50:45] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:50:45] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412245@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:50:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:50:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:50:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:50:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:50:45] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:50:45] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:38499 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=38499 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as512ee78f Call-ID: 1344412245@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:50:45] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:38499 [Aug 8 09:50:45] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412245@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:50:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412215@127.0.0.1' [Aug 8 09:50:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412215@127.0.0.1 [Aug 8 09:50:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412215@127.0.0.1' Method: OPTIONS [Aug 8 09:50:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:50:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:50:55] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:56177 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412255@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412255@127.0.0.1 [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:50:55] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:50:55] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:50:55] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412255@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:50:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:50:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:50:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:50:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:50:55] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:50:55] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:56177 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=56177 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as0e97e9ae Call-ID: 1344412255@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:50:55] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:56177 [Aug 8 09:50:55] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412255@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:50:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412225@127.0.0.1' [Aug 8 09:50:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412225@127.0.0.1 [Aug 8 09:50:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412225@127.0.0.1' Method: OPTIONS [Aug 8 09:51:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:51:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:42268 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412265@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412265@127.0.0.1 [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:51:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:51:05] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:51:05] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412265@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:51:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:51:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:51:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:51:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:51:05] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:51:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:42268 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=42268 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as18d528c7 Call-ID: 1344412265@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:51:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:42268 [Aug 8 09:51:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412265@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:51:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412235@127.0.0.1' [Aug 8 09:51:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412235@127.0.0.1 [Aug 8 09:51:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412235@127.0.0.1' Method: OPTIONS [Aug 8 09:51:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:51:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:15] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:42396 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412275@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412275@127.0.0.1 [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:51:15] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:51:15] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:51:15] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412275@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:51:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:51:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:51:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:51:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:51:15] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:51:15] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:42396 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=42396 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as1968d0fc Call-ID: 1344412275@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:51:15] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:42396 [Aug 8 09:51:15] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412275@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:51:17] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412245@127.0.0.1' [Aug 8 09:51:17] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412245@127.0.0.1 [Aug 8 09:51:17] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412245@127.0.0.1' Method: OPTIONS [Aug 8 09:51:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:51:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:25] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:33855 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412285@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412285@127.0.0.1 [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:51:25] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:51:25] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:51:25] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412285@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:51:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:51:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:51:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:51:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:51:25] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:51:25] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:33855 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=33855 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as3f1dc537 Call-ID: 1344412285@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:51:25] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:33855 [Aug 8 09:51:25] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412285@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:51:27] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412255@127.0.0.1' [Aug 8 09:51:27] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412255@127.0.0.1 [Aug 8 09:51:27] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412255@127.0.0.1' Method: OPTIONS [Aug 8 09:51:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 59077b16550b4e081c8eabe949537b5e@(null) - OPTIONS (No RTP) [Aug 8 09:51:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:51:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 0e04c5de13bd696d22c1d0111262641a@127.0.0.1:5060 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7413124d [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as75eafb00 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 0e04c5de13bd696d22c1d0111262641a@127.0.0.1:5060 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:51:32 GMT [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:51:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7413124d Max-Forwards: 70 From: "Unknown" ;tag=as75eafb00 To: Contact: Call-ID: 0e04c5de13bd696d22c1d0111262641a@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:51:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #426 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:51:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7413124d Max-Forwards: 70 From: "Unknown" ;tag=as75eafb00 To: Contact: Call-ID: 0e04c5de13bd696d22c1d0111262641a@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:51:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7413124d [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as75eafb00 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 0e04c5de13bd696d22c1d0111262641a@127.0.0.1:5060 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:51:32 GMT [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:51:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:51:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:51:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:51:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:51:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:51:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:51:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7413124d;received=127.0.0.1 From: "Unknown" ;tag=as75eafb00 To: ;tag=as75eafb00 Call-ID: 0e04c5de13bd696d22c1d0111262641a@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:51:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '0e04c5de13bd696d22c1d0111262641a@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:51:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7413124d;received=127.0.0.1 From: "Unknown" ;tag=as75eafb00 To: ;tag=as75eafb00 Call-ID: 0e04c5de13bd696d22c1d0111262641a@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7413124d;received=127.0.0.1 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as75eafb00 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as75eafb00 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 0e04c5de13bd696d22c1d0111262641a@127.0.0.1:5060 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:51:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #426 [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '0e04c5de13bd696d22c1d0111262641a@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:51:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 0e04c5de13bd696d22c1d0111262641a@127.0.0.1:5060 [Aug 8 09:51:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '0e04c5de13bd696d22c1d0111262641a@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:51:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:51:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:35] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:46586 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412295@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412295@127.0.0.1 [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:51:35] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:51:35] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:51:35] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412295@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:51:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:51:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:51:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:51:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:51:35] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:51:35] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:46586 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=46586 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as6c11fb62 Call-ID: 1344412295@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:51:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:46586 [Aug 8 09:51:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412295@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:51:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412265@127.0.0.1' [Aug 8 09:51:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412265@127.0.0.1 [Aug 8 09:51:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412265@127.0.0.1' Method: OPTIONS [Aug 8 09:51:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:51:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:45] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:41928 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412305@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412305@127.0.0.1 [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:51:45] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:51:45] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:51:45] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412305@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:51:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:51:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:51:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:51:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:51:45] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:51:45] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:41928 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=41928 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as47ef2bd2 Call-ID: 1344412305@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:51:45] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:41928 [Aug 8 09:51:45] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412305@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:51:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412275@127.0.0.1' [Aug 8 09:51:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412275@127.0.0.1 [Aug 8 09:51:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412275@127.0.0.1' Method: OPTIONS [Aug 8 09:51:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:52] DEBUG[2982] manager.c: Running action 'IAXpeers' [Aug 8 09:51:52] DEBUG[2982] manager.c: Running action 'SIPpeers' [Aug 8 09:51:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:51:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:51:55] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:58724 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412315@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412315@127.0.0.1 [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:51:55] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:51:55] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:51:55] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412315@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:51:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:51:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:51:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:51:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:51:55] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:51:55] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:58724 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=58724 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as26be15f2 Call-ID: 1344412315@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:51:55] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:58724 [Aug 8 09:51:55] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412315@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:51:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412285@127.0.0.1' [Aug 8 09:51:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412285@127.0.0.1 [Aug 8 09:51:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412285@127.0.0.1' Method: OPTIONS [Aug 8 09:52:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:52:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:33764 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412325@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412325@127.0.0.1 [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:52:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:52:05] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:52:05] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412325@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:52:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:52:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:52:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:52:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:52:05] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:52:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:33764 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=33764 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as16c07589 Call-ID: 1344412325@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:52:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:33764 [Aug 8 09:52:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412325@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:52:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412295@127.0.0.1' [Aug 8 09:52:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412295@127.0.0.1 [Aug 8 09:52:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412295@127.0.0.1' Method: OPTIONS [Aug 8 09:52:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:52:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:15] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:47252 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412335@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412335@127.0.0.1 [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:52:15] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:52:15] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:52:15] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412335@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:52:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:52:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:52:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:52:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:52:15] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:52:15] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:47252 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=47252 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as662134bb Call-ID: 1344412335@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:52:15] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:47252 [Aug 8 09:52:15] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412335@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:52:17] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412305@127.0.0.1' [Aug 8 09:52:17] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412305@127.0.0.1 [Aug 8 09:52:17] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412305@127.0.0.1' Method: OPTIONS [Aug 8 09:52:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:52:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:25] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:49501 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412345@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412345@127.0.0.1 [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:52:25] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:52:25] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:52:25] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412345@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:52:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:52:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:52:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:52:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:52:25] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:52:25] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:49501 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=49501 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as47d9fb56 Call-ID: 1344412345@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:52:25] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:49501 [Aug 8 09:52:25] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412345@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:52:27] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412315@127.0.0.1' [Aug 8 09:52:27] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412315@127.0.0.1 [Aug 8 09:52:27] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412315@127.0.0.1' Method: OPTIONS [Aug 8 09:52:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 24d885362b8d60434062188a70d0fe4f@(null) - OPTIONS (No RTP) [Aug 8 09:52:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:52:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 7636e2a02f49a42638542c882ad889b6@127.0.0.1:5060 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK42d71a63 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as52bf7db5 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 7636e2a02f49a42638542c882ad889b6@127.0.0.1:5060 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:52:32 GMT [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:52:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK42d71a63 Max-Forwards: 70 From: "Unknown" ;tag=as52bf7db5 To: Contact: Call-ID: 7636e2a02f49a42638542c882ad889b6@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:52:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #436 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:52:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK42d71a63 Max-Forwards: 70 From: "Unknown" ;tag=as52bf7db5 To: Contact: Call-ID: 7636e2a02f49a42638542c882ad889b6@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:52:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK42d71a63 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as52bf7db5 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 7636e2a02f49a42638542c882ad889b6@127.0.0.1:5060 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:52:32 GMT [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:52:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:52:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:52:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:52:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:52:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:52:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:52:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK42d71a63;received=127.0.0.1 From: "Unknown" ;tag=as52bf7db5 To: ;tag=as52bf7db5 Call-ID: 7636e2a02f49a42638542c882ad889b6@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:52:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '7636e2a02f49a42638542c882ad889b6@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:52:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK42d71a63;received=127.0.0.1 From: "Unknown" ;tag=as52bf7db5 To: ;tag=as52bf7db5 Call-ID: 7636e2a02f49a42638542c882ad889b6@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK42d71a63;received=127.0.0.1 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as52bf7db5 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as52bf7db5 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 7636e2a02f49a42638542c882ad889b6@127.0.0.1:5060 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:52:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #436 [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '7636e2a02f49a42638542c882ad889b6@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:52:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 7636e2a02f49a42638542c882ad889b6@127.0.0.1:5060 [Aug 8 09:52:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '7636e2a02f49a42638542c882ad889b6@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:52:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:52:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:35] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:46937 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412355@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412355@127.0.0.1 [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:52:35] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:52:35] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:52:35] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412355@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:52:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:52:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:52:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:52:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:52:35] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:52:35] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:46937 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=46937 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as7c11317a Call-ID: 1344412355@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:52:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:46937 [Aug 8 09:52:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412355@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:52:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412325@127.0.0.1' [Aug 8 09:52:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412325@127.0.0.1 [Aug 8 09:52:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412325@127.0.0.1' Method: OPTIONS [Aug 8 09:52:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:52:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:45] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:50611 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412365@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412365@127.0.0.1 [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:52:45] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:52:45] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:52:45] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412365@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:52:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:52:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:52:45] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:52:45] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:52:45] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:52:45] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:50611 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=50611 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as49eb042e Call-ID: 1344412365@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:52:45] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:50611 [Aug 8 09:52:45] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412365@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:52:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412335@127.0.0.1' [Aug 8 09:52:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412335@127.0.0.1 [Aug 8 09:52:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412335@127.0.0.1' Method: OPTIONS [Aug 8 09:52:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:52:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:52:55] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:48077 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412375@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412375@127.0.0.1 [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:52:55] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:52:55] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:52:55] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412375@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:52:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:52:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:52:55] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:52:55] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:52:55] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:52:55] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:48077 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=48077 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as120c6158 Call-ID: 1344412375@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:52:55] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:48077 [Aug 8 09:52:55] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412375@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:52:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412345@127.0.0.1' [Aug 8 09:52:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412345@127.0.0.1 [Aug 8 09:52:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412345@127.0.0.1' Method: OPTIONS [Aug 8 09:53:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:53:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:58136 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412385@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412385@127.0.0.1 [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:53:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:53:05] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:53:05] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412385@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:53:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:53:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:53:05] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:53:05] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:53:05] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:53:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:58136 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=58136 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as44771c1d Call-ID: 1344412385@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:53:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:58136 [Aug 8 09:53:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412385@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:53:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412355@127.0.0.1' [Aug 8 09:53:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412355@127.0.0.1 [Aug 8 09:53:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412355@127.0.0.1' Method: OPTIONS [Aug 8 09:53:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:53:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:15] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:34777 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412395@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412395@127.0.0.1 [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:53:15] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:53:15] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:53:15] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412395@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:53:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:53:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:53:15] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:53:15] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:53:15] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:53:15] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:34777 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=34777 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as00b1238e Call-ID: 1344412395@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:53:15] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:34777 [Aug 8 09:53:15] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412395@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:53:17] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412365@127.0.0.1' [Aug 8 09:53:17] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412365@127.0.0.1 [Aug 8 09:53:17] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412365@127.0.0.1' Method: OPTIONS [Aug 8 09:53:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:53:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:25] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:39716 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412405@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412405@127.0.0.1 [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:53:25] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:53:25] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:53:25] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412405@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:53:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:53:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:53:25] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:53:25] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:53:25] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:53:25] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:39716 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=39716 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as6b46381d Call-ID: 1344412405@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:53:25] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:39716 [Aug 8 09:53:25] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412405@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:53:27] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412375@127.0.0.1' [Aug 8 09:53:27] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412375@127.0.0.1 [Aug 8 09:53:27] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412375@127.0.0.1' Method: OPTIONS [Aug 8 09:53:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 68e892684814d853309f18b7678712f8@(null) - OPTIONS (No RTP) [Aug 8 09:53:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:53:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 32d0366802c5080c73a6373c6c0802fc@127.0.0.1:5060 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK66104b8b [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as618d0813 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 32d0366802c5080c73a6373c6c0802fc@127.0.0.1:5060 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:53:32 GMT [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:53:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK66104b8b Max-Forwards: 70 From: "Unknown" ;tag=as618d0813 To: Contact: Call-ID: 32d0366802c5080c73a6373c6c0802fc@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:53:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #446 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:53:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK66104b8b Max-Forwards: 70 From: "Unknown" ;tag=as618d0813 To: Contact: Call-ID: 32d0366802c5080c73a6373c6c0802fc@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:53:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK66104b8b [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as618d0813 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 32d0366802c5080c73a6373c6c0802fc@127.0.0.1:5060 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:53:32 GMT [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:53:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:53:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:53:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:53:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:53:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:53:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:53:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK66104b8b;received=127.0.0.1 From: "Unknown" ;tag=as618d0813 To: ;tag=as618d0813 Call-ID: 32d0366802c5080c73a6373c6c0802fc@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:53:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '32d0366802c5080c73a6373c6c0802fc@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:53:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK66104b8b;received=127.0.0.1 From: "Unknown" ;tag=as618d0813 To: ;tag=as618d0813 Call-ID: 32d0366802c5080c73a6373c6c0802fc@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK66104b8b;received=127.0.0.1 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as618d0813 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as618d0813 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 32d0366802c5080c73a6373c6c0802fc@127.0.0.1:5060 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:53:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #446 [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '32d0366802c5080c73a6373c6c0802fc@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:53:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 32d0366802c5080c73a6373c6c0802fc@127.0.0.1:5060 [Aug 8 09:53:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '32d0366802c5080c73a6373c6c0802fc@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:53:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:53:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:35] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:53993 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412415@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412415@127.0.0.1 [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:53:35] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:53:35] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:53:35] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412415@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:53:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:53:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:53:35] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:53:35] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:53:35] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:53:35] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:53993 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=53993 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as460dfcc9 Call-ID: 1344412415@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:53:35] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:53993 [Aug 8 09:53:35] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412415@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:53:37] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412385@127.0.0.1' [Aug 8 09:53:37] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412385@127.0.0.1 [Aug 8 09:53:37] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412385@127.0.0.1' Method: OPTIONS [Aug 8 09:53:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:53:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:46] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:56402 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412425@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412425@127.0.0.1 [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:53:46] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:53:46] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:53:46] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412425@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:53:46] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:53:46] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:53:46] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:53:46] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:53:46] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:53:46] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:56402 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=56402 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as154829a6 Call-ID: 1344412425@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:53:46] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:56402 [Aug 8 09:53:46] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412425@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:53:47] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412395@127.0.0.1' [Aug 8 09:53:47] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412395@127.0.0.1 [Aug 8 09:53:47] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412395@127.0.0.1' Method: OPTIONS [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Session timer expired: 288 - 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Strict routing enforced for session 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:53:48] VERBOSE[2371] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 8 09:53:48] DEBUG[2371] netsock2.c: Splitting '192.168.1.102:2052' into... [Aug 8 09:53:48] DEBUG[2371] netsock2.c: ...host '192.168.1.102' and port '2052'. [Aug 8 09:53:48] VERBOSE[2371] chan_sip.c: set_destination: set destination to 192.168.1.102:2052 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Aug 8 09:53:48] VERBOSE[2371] chan_sip.c: Audio is at 18588 [Aug 8 09:53:48] VERBOSE[2371] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 8 09:53:48] VERBOSE[2371] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 8 09:53:48] VERBOSE[2371] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: -- Done with adding codecs to SDP [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Initializing already initialized SIP dialog 3c39db8cc1a5-pv4xf5eh5w4v (presumably reinvite) [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 0 [ 56]: INVITE sip:2209@192.168.1.102:2052;line=882y5m72 SIP/2.0 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK08867819;rport [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 3 [ 45]: From: ;tag=as0bbc9cb2 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 4 [ 50]: To: "2209" ;tag=gbdfdvpkud [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 5 [ 38]: Contact: [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 6 [ 34]: Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 13 [ 47]: X-asterisk-Info: SIP re-invite (Session-Timers) [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Aug 8 09:53:48] VERBOSE[2371] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.102:2052: INVITE sip:2209@192.168.1.102:2052;line=882y5m72 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK08867819;rport Max-Forwards: 70 From: ;tag=as0bbc9cb2 To: "2209" ;tag=gbdfdvpkud Contact: Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v CSeq: 102 INVITE User-Agent: asterisk Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 333 v=0 o=tvox 842690090 842690090 IN IP4 192.168.0.178 s=asterisk c=IN IP4 192.168.0.178 t=0 0 m=audio 18588 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:CStJb7zHpMiDirfLQN85uIIeE/fGvo9zKi69GowG --- [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #452 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.1.102:2052 [Aug 8 09:53:48] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2052 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK08867819;rport=5060 From: ;tag=as0bbc9cb2 To: "2209" ;tag=gbdfdvpkud Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v CSeq: 102 INVITE Contact: ;reg-id=1 Require: timer Session-Expires: 1800;refresher=uas User-Agent: snom300/8.4.32 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 331 v=0 o=root 1987219225 1987219225 IN IP4 192.168.1.102 s=call c=IN IP4 192.168.1.102 t=0 0 m=audio 11460 RTP/SAVP 8 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:JATbmi5O79v44WmAN8Q/Ad5xGjCkgnmjDLf5IdvF a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK08867819;rport=5060 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 2 [ 45]: From: ;tag=as0bbc9cb2 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 3 [ 50]: To: "2209" ;tag=gbdfdvpkud [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 4 [ 34]: Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 7 [ 14]: Require: timer [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 8 [ 35]: Session-Expires: 1800;refresher=uas [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 9 [ 26]: User-Agent: snom300/8.4.32 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 14 [ 19]: Content-Length: 331 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Header 15 [ 0]: [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Body 0 [ 3]: v=0 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Body 1 [ 49]: o=root 1987219225 1987219225 IN IP4 192.168.1.102 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Body 2 [ 6]: s=call [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.1.102 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Body 5 [ 30]: m=audio 11460 RTP/SAVP 8 0 101 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:JATbmi5O79v44WmAN8Q/Ad5xGjCkgnmjDLf5IdvF [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Body 12 [ 10]: a=sendrecv [Aug 8 09:53:48] VERBOSE[2371] chan_sip.c: --- (15 headers 13 lines) --- [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Acked pending invite 102 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #452 [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Stopping retransmission on '3c39db8cc1a5-pv4xf5eh5w4v' of Request 102: Match Found [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: SIP response 200 to RE-invite on outgoing call 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Call 3c39db8cc1a5-pv4xf5eh5w4v responded to our reinvite without changing SDP version; ignoring SDP. [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Updating call counter for incoming call [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Strict routing enforced for session 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:53:48] VERBOSE[2371] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 8 09:53:48] DEBUG[2371] netsock2.c: Splitting '192.168.1.102:2052' into... [Aug 8 09:53:48] DEBUG[2371] netsock2.c: ...host '192.168.1.102' and port '2052'. [Aug 8 09:53:48] VERBOSE[2371] chan_sip.c: set_destination: set destination to 192.168.1.102:2052 [Aug 8 09:53:48] VERBOSE[2371] chan_sip.c: Transmitting (NAT) to 192.168.1.102:2052: ACK sip:2209@192.168.1.102:2052;line=882y5m72 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK0a866557;rport Max-Forwards: 70 From: ;tag=as0bbc9cb2 To: "2209" ;tag=gbdfdvpkud Contact: Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v CSeq: 102 ACK User-Agent: asterisk Content-Length: 0 --- [Aug 8 09:53:48] DEBUG[2371] chan_sip.c: Trying to put 'ACK sip:220' onto UDP socket destined for 192.168.1.102:2052 [Aug 8 09:53:48] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2209 [Aug 8 09:53:48] DEBUG[2321] chan_sip.c: Checking device state for peer 2209 [Aug 8 09:53:48] DEBUG[2321] devicestate.c: Changing state for SIP/2209 - state 2 (In use) [Aug 8 09:53:48] DEBUG[2321] devicestate.c: device 'SIP/2209' state '2' [Aug 8 09:53:48] DEBUG[5008] channel.c: Got a FRAME_CONTROL (31) frame on channel SIP/2209-00000000 [Aug 8 09:53:48] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:53:48] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:53:48] DEBUG[5008] channel.c: Bridge stops bridging channels SIP/2209-00000000 and SIP/2210-00000001 [Aug 8 09:53:48] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:53:48] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:53:48] DEBUG[2980] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:53:48] DEBUG[2980] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:53:48] DEBUG[2984] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:53:48] DEBUG[2984] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:53:48] DEBUG[2988] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:53:48] DEBUG[2988] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:53:48] DEBUG[3176] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:53:48] DEBUG[3176] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:53:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPEER Value: SIP/2210-00000001 Uniqueid: 1344411528.0 [Aug 8 09:53:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPVTCALLID Value: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 Uniqueid: 1344411528.0 [Aug 8 09:53:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPEER Value: SIP/2209-00000000 Uniqueid: 1344411528.1 [Aug 8 09:53:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPVTCALLID Value: 3c39db8cc1a5-pv4xf5eh5w4v Uniqueid: 1344411528.1 [Aug 8 09:53:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPEER Value: SIP/2210-00000001 Uniqueid: 1344411528.0 [Aug 8 09:53:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPVTCALLID Value: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 Uniqueid: 1344411528.0 [Aug 8 09:53:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPEER Value: SIP/2209-00000000 Uniqueid: 1344411528.1 [Aug 8 09:53:48] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPVTCALLID Value: 3c39db8cc1a5-pv4xf5eh5w4v Uniqueid: 1344411528.1 [Aug 8 09:53:48] DEBUG[2394] app_queue.c: Device 'SIP/2209' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 8 09:53:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:53:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:53:56] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:38158 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412436@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412436@127.0.0.1 [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:53:56] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:53:56] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:53:56] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412436@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:53:56] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:53:56] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:53:56] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:53:56] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:53:56] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:53:56] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:38158 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=38158 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as66d7cfd8 Call-ID: 1344412436@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:53:56] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:38158 [Aug 8 09:53:56] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412436@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:53:57] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412405@127.0.0.1' [Aug 8 09:53:57] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412405@127.0.0.1 [Aug 8 09:53:57] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412405@127.0.0.1' Method: OPTIONS [Aug 8 09:54:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:54:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:06] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:52841 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412446@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412446@127.0.0.1 [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:54:06] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:54:06] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:54:06] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412446@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:54:06] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:54:06] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:54:06] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:54:06] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:54:06] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:54:06] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:52841 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=52841 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as6f7b83a7 Call-ID: 1344412446@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:54:06] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:52841 [Aug 8 09:54:06] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412446@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:54:07] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412415@127.0.0.1' [Aug 8 09:54:07] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412415@127.0.0.1 [Aug 8 09:54:07] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412415@127.0.0.1' Method: OPTIONS [Aug 8 09:54:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:54:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:16] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:51753 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412456@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412456@127.0.0.1 [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:54:16] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:54:16] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:54:16] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412456@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:54:16] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:54:16] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:54:16] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:54:16] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:54:16] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:54:16] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:51753 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=51753 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as550071d9 Call-ID: 1344412456@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:54:16] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:51753 [Aug 8 09:54:16] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412456@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:54:18] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412425@127.0.0.1' [Aug 8 09:54:18] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412425@127.0.0.1 [Aug 8 09:54:18] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412425@127.0.0.1' Method: OPTIONS [Aug 8 09:54:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:54:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:26] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:44766 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412466@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412466@127.0.0.1 [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:54:26] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:54:26] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:54:26] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412466@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:54:26] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:54:26] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:54:26] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:54:26] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:54:26] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:54:26] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:44766 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=44766 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as45dd0dcd Call-ID: 1344412466@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:54:26] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:44766 [Aug 8 09:54:26] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412466@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:54:28] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412436@127.0.0.1' [Aug 8 09:54:28] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412436@127.0.0.1 [Aug 8 09:54:28] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412436@127.0.0.1' Method: OPTIONS [Aug 8 09:54:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 2c7789e17f149d4a7767d4e051d7880a@(null) - OPTIONS (No RTP) [Aug 8 09:54:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:54:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 4826fd425d2692f42297217711396c0d@127.0.0.1:5060 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK31b98cb2 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as6f226e50 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 4826fd425d2692f42297217711396c0d@127.0.0.1:5060 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:54:32 GMT [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:54:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK31b98cb2 Max-Forwards: 70 From: "Unknown" ;tag=as6f226e50 To: Contact: Call-ID: 4826fd425d2692f42297217711396c0d@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:54:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #457 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:54:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK31b98cb2 Max-Forwards: 70 From: "Unknown" ;tag=as6f226e50 To: Contact: Call-ID: 4826fd425d2692f42297217711396c0d@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:54:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK31b98cb2 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as6f226e50 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 4826fd425d2692f42297217711396c0d@127.0.0.1:5060 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:54:32 GMT [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:54:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:54:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:54:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:54:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:54:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:54:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:54:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK31b98cb2;received=127.0.0.1 From: "Unknown" ;tag=as6f226e50 To: ;tag=as6f226e50 Call-ID: 4826fd425d2692f42297217711396c0d@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:54:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '4826fd425d2692f42297217711396c0d@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:54:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK31b98cb2;received=127.0.0.1 From: "Unknown" ;tag=as6f226e50 To: ;tag=as6f226e50 Call-ID: 4826fd425d2692f42297217711396c0d@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK31b98cb2;received=127.0.0.1 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as6f226e50 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as6f226e50 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 4826fd425d2692f42297217711396c0d@127.0.0.1:5060 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:54:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #457 [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '4826fd425d2692f42297217711396c0d@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:54:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 4826fd425d2692f42297217711396c0d@127.0.0.1:5060 [Aug 8 09:54:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '4826fd425d2692f42297217711396c0d@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:54:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:54:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:36] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:40886 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412476@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412476@127.0.0.1 [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:54:36] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:54:36] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:54:36] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412476@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:54:36] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:54:36] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:54:36] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:54:36] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:54:36] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:54:36] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:40886 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=40886 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as3ec3dd92 Call-ID: 1344412476@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:54:36] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:40886 [Aug 8 09:54:36] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412476@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:54:38] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412446@127.0.0.1' [Aug 8 09:54:38] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412446@127.0.0.1 [Aug 8 09:54:38] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412446@127.0.0.1' Method: OPTIONS [Aug 8 09:54:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:54:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:46] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:49664 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412486@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412486@127.0.0.1 [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:54:46] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:54:46] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:54:46] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412486@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:54:46] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:54:46] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:54:46] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:54:46] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:54:46] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:54:46] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:49664 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=49664 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as14c9bd22 Call-ID: 1344412486@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:54:46] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:49664 [Aug 8 09:54:46] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412486@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:54:48] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412456@127.0.0.1' [Aug 8 09:54:48] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412456@127.0.0.1 [Aug 8 09:54:48] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412456@127.0.0.1' Method: OPTIONS [Aug 8 09:54:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:54:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:54:56] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:51778 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412496@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412496@127.0.0.1 [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:54:56] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:54:56] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:54:56] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412496@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:54:56] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:54:56] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:54:56] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:54:56] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:54:56] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:54:56] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:51778 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=51778 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as4dfa023f Call-ID: 1344412496@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:54:56] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:51778 [Aug 8 09:54:56] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412496@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:54:58] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412466@127.0.0.1' [Aug 8 09:54:58] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412466@127.0.0.1 [Aug 8 09:54:58] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412466@127.0.0.1' Method: OPTIONS [Aug 8 09:55:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:55:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:06] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:54728 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412506@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412506@127.0.0.1 [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:55:06] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:55:06] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:55:06] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412506@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:55:06] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:55:06] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:55:06] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:55:06] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:55:06] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:55:06] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:54728 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=54728 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as0e94746a Call-ID: 1344412506@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:55:06] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:54728 [Aug 8 09:55:06] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412506@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:55:08] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412476@127.0.0.1' [Aug 8 09:55:08] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412476@127.0.0.1 [Aug 8 09:55:08] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412476@127.0.0.1' Method: OPTIONS [Aug 8 09:55:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:55:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:16] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:38521 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412516@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412516@127.0.0.1 [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:55:16] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:55:16] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:55:16] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412516@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:55:16] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:55:16] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:55:16] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:55:16] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:55:16] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:55:16] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:38521 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=38521 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as6f790ebe Call-ID: 1344412516@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:55:16] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:38521 [Aug 8 09:55:16] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412516@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:55:18] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412486@127.0.0.1' [Aug 8 09:55:18] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412486@127.0.0.1 [Aug 8 09:55:18] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412486@127.0.0.1' Method: OPTIONS [Aug 8 09:55:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:55:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:26] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:48696 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412526@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412526@127.0.0.1 [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:55:26] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:55:26] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:55:26] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412526@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:55:26] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:55:26] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:55:26] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:55:26] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:55:26] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:55:26] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:48696 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=48696 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as354843cb Call-ID: 1344412526@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:55:26] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:48696 [Aug 8 09:55:26] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412526@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:55:28] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412496@127.0.0.1' [Aug 8 09:55:28] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412496@127.0.0.1 [Aug 8 09:55:28] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412496@127.0.0.1' Method: OPTIONS [Aug 8 09:55:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 6149672e68a7f7721138bf112f10a8cc@(null) - OPTIONS (No RTP) [Aug 8 09:55:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:55:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 640d6be44b3b0f6f620440a178e5ee69@127.0.0.1:5060 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK00b0c37f [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as5697fb05 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 640d6be44b3b0f6f620440a178e5ee69@127.0.0.1:5060 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:55:32 GMT [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:55:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK00b0c37f Max-Forwards: 70 From: "Unknown" ;tag=as5697fb05 To: Contact: Call-ID: 640d6be44b3b0f6f620440a178e5ee69@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:55:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #467 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:55:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK00b0c37f Max-Forwards: 70 From: "Unknown" ;tag=as5697fb05 To: Contact: Call-ID: 640d6be44b3b0f6f620440a178e5ee69@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:55:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK00b0c37f [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as5697fb05 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 640d6be44b3b0f6f620440a178e5ee69@127.0.0.1:5060 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:55:32 GMT [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:55:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:55:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:55:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:55:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:55:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:55:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:55:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK00b0c37f;received=127.0.0.1 From: "Unknown" ;tag=as5697fb05 To: ;tag=as5697fb05 Call-ID: 640d6be44b3b0f6f620440a178e5ee69@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:55:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '640d6be44b3b0f6f620440a178e5ee69@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:55:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK00b0c37f;received=127.0.0.1 From: "Unknown" ;tag=as5697fb05 To: ;tag=as5697fb05 Call-ID: 640d6be44b3b0f6f620440a178e5ee69@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK00b0c37f;received=127.0.0.1 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as5697fb05 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as5697fb05 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 640d6be44b3b0f6f620440a178e5ee69@127.0.0.1:5060 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:55:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #467 [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '640d6be44b3b0f6f620440a178e5ee69@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:55:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 640d6be44b3b0f6f620440a178e5ee69@127.0.0.1:5060 [Aug 8 09:55:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '640d6be44b3b0f6f620440a178e5ee69@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:55:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:55:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:36] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:50165 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412536@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412536@127.0.0.1 [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:55:36] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:55:36] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:55:36] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412536@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:55:36] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:55:36] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:55:36] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:55:36] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:55:36] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:55:36] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:50165 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=50165 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as138c8f74 Call-ID: 1344412536@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:55:36] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:50165 [Aug 8 09:55:36] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412536@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:55:38] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412506@127.0.0.1' [Aug 8 09:55:38] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412506@127.0.0.1 [Aug 8 09:55:38] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412506@127.0.0.1' Method: OPTIONS [Aug 8 09:55:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:55:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:46] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:60555 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412546@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412546@127.0.0.1 [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:55:46] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:55:46] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:55:46] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412546@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:55:46] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:55:46] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:55:46] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:55:46] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:55:46] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:55:46] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:60555 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=60555 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as3519909b Call-ID: 1344412546@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:55:46] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:60555 [Aug 8 09:55:46] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412546@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:55:48] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412516@127.0.0.1' [Aug 8 09:55:48] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412516@127.0.0.1 [Aug 8 09:55:48] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412516@127.0.0.1' Method: OPTIONS [Aug 8 09:55:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:55:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:55:56] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:51589 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412556@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412556@127.0.0.1 [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:55:56] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:55:56] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:55:56] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412556@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:55:56] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:55:56] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:55:56] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:55:56] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:55:56] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:55:56] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:51589 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=51589 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as140d2e42 Call-ID: 1344412556@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:55:56] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:51589 [Aug 8 09:55:56] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412556@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:55:58] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412526@127.0.0.1' [Aug 8 09:55:58] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412526@127.0.0.1 [Aug 8 09:55:58] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412526@127.0.0.1' Method: OPTIONS [Aug 8 09:56:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:56:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:06] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:41510 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412566@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412566@127.0.0.1 [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:56:06] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:56:06] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:56:06] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412566@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:56:06] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:56:06] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:56:06] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:56:06] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:56:06] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:56:06] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:41510 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=41510 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as653056ac Call-ID: 1344412566@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:56:06] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:41510 [Aug 8 09:56:06] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412566@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:56:08] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412536@127.0.0.1' [Aug 8 09:56:08] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412536@127.0.0.1 [Aug 8 09:56:08] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412536@127.0.0.1' Method: OPTIONS [Aug 8 09:56:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:56:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:16] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:35013 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412576@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412576@127.0.0.1 [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:56:16] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:56:16] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:56:16] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412576@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:56:16] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:56:16] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:56:16] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:56:16] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:56:16] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:56:16] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:35013 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=35013 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as44944f07 Call-ID: 1344412576@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:56:16] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:35013 [Aug 8 09:56:16] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412576@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:56:18] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412546@127.0.0.1' [Aug 8 09:56:18] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412546@127.0.0.1 [Aug 8 09:56:18] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412546@127.0.0.1' Method: OPTIONS [Aug 8 09:56:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:56:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:26] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:60463 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412586@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412586@127.0.0.1 [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:56:26] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:56:26] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:56:26] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412586@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:56:26] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:56:26] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:56:26] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:56:26] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:56:26] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:56:26] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:60463 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=60463 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as571183de Call-ID: 1344412586@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:56:26] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:60463 [Aug 8 09:56:26] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412586@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:56:28] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412556@127.0.0.1' [Aug 8 09:56:28] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412556@127.0.0.1 [Aug 8 09:56:28] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412556@127.0.0.1' Method: OPTIONS [Aug 8 09:56:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 60266c85360d2b8113f08ba342172799@(null) - OPTIONS (No RTP) [Aug 8 09:56:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:56:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 04666715551a8aea40af78691d59202a@127.0.0.1:5060 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK568d9868 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as7de1318a [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 04666715551a8aea40af78691d59202a@127.0.0.1:5060 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:56:32 GMT [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:56:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK568d9868 Max-Forwards: 70 From: "Unknown" ;tag=as7de1318a To: Contact: Call-ID: 04666715551a8aea40af78691d59202a@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:56:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #477 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:56:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK568d9868 Max-Forwards: 70 From: "Unknown" ;tag=as7de1318a To: Contact: Call-ID: 04666715551a8aea40af78691d59202a@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:56:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK568d9868 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as7de1318a [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 04666715551a8aea40af78691d59202a@127.0.0.1:5060 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:56:32 GMT [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:56:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:56:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:56:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:56:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:56:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:56:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:56:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK568d9868;received=127.0.0.1 From: "Unknown" ;tag=as7de1318a To: ;tag=as7de1318a Call-ID: 04666715551a8aea40af78691d59202a@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:56:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '04666715551a8aea40af78691d59202a@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:56:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK568d9868;received=127.0.0.1 From: "Unknown" ;tag=as7de1318a To: ;tag=as7de1318a Call-ID: 04666715551a8aea40af78691d59202a@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK568d9868;received=127.0.0.1 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as7de1318a [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as7de1318a [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 04666715551a8aea40af78691d59202a@127.0.0.1:5060 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:56:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #477 [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '04666715551a8aea40af78691d59202a@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:56:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 04666715551a8aea40af78691d59202a@127.0.0.1:5060 [Aug 8 09:56:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '04666715551a8aea40af78691d59202a@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:56:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:56:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:36] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:39479 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412596@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412596@127.0.0.1 [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:56:36] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:56:36] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:56:36] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412596@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:56:36] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:56:36] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:56:36] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:56:36] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:56:36] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:56:36] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:39479 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=39479 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as10fb455a Call-ID: 1344412596@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:56:36] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:39479 [Aug 8 09:56:36] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412596@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:56:38] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412566@127.0.0.1' [Aug 8 09:56:38] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412566@127.0.0.1 [Aug 8 09:56:38] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412566@127.0.0.1' Method: OPTIONS [Aug 8 09:56:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:56:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:46] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:53345 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412606@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412606@127.0.0.1 [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:56:46] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:56:46] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:56:46] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412606@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:56:46] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:56:46] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:56:46] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:56:46] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:56:46] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:56:46] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:53345 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=53345 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as090bf417 Call-ID: 1344412606@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:56:46] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:53345 [Aug 8 09:56:46] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412606@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:56:48] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412576@127.0.0.1' [Aug 8 09:56:48] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412576@127.0.0.1 [Aug 8 09:56:48] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412576@127.0.0.1' Method: OPTIONS [Aug 8 09:56:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:52] DEBUG[2982] manager.c: Running action 'IAXpeers' [Aug 8 09:56:52] DEBUG[2982] manager.c: Running action 'SIPpeers' [Aug 8 09:56:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:56:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:56:56] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:57525 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412616@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412616@127.0.0.1 [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:56:56] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:56:56] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:56:56] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412616@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:56:56] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:56:56] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:56:56] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:56:56] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:56:56] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:56:56] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:57525 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=57525 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as010c824a Call-ID: 1344412616@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:56:56] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:57525 [Aug 8 09:56:56] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412616@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:56:58] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412586@127.0.0.1' [Aug 8 09:56:58] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412586@127.0.0.1 [Aug 8 09:56:58] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412586@127.0.0.1' Method: OPTIONS [Aug 8 09:57:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:57:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:06] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:39373 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412626@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412626@127.0.0.1 [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:57:06] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:57:06] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:57:06] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412626@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:57:06] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:57:06] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:57:06] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:57:06] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:57:06] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:57:06] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:39373 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=39373 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as0f441ab9 Call-ID: 1344412626@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:57:06] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:39373 [Aug 8 09:57:06] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412626@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:57:08] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412596@127.0.0.1' [Aug 8 09:57:08] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412596@127.0.0.1 [Aug 8 09:57:08] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412596@127.0.0.1' Method: OPTIONS [Aug 8 09:57:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:57:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:16] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:41992 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412636@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412636@127.0.0.1 [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:57:16] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:57:16] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:57:16] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412636@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:57:16] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:57:16] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:57:16] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:57:16] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:57:16] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:57:16] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:41992 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=41992 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as7f26559a Call-ID: 1344412636@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:57:16] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:41992 [Aug 8 09:57:16] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412636@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:57:18] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412606@127.0.0.1' [Aug 8 09:57:18] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412606@127.0.0.1 [Aug 8 09:57:18] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412606@127.0.0.1' Method: OPTIONS [Aug 8 09:57:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:20] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:57:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:25] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:26] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:44146 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412646@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412646@127.0.0.1 [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:57:26] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:57:26] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:57:26] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412646@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:57:26] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:57:26] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:57:26] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:57:26] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:57:26] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:57:26] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:44146 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=44146 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as04f99704 Call-ID: 1344412646@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:57:26] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:44146 [Aug 8 09:57:26] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412646@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:57:28] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412616@127.0.0.1' [Aug 8 09:57:28] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412616@127.0.0.1 [Aug 8 09:57:28] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412616@127.0.0.1' Method: OPTIONS [Aug 8 09:57:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:30] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 64d3ca4d75c95c1e75ed528e3d450369@(null) - OPTIONS (No RTP) [Aug 8 09:57:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:57:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 6b11e76002ce6829560774012820a94b@127.0.0.1:5060 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK626d735b [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as71663e69 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 6b11e76002ce6829560774012820a94b@127.0.0.1:5060 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:57:32 GMT [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:57:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK626d735b Max-Forwards: 70 From: "Unknown" ;tag=as71663e69 To: Contact: Call-ID: 6b11e76002ce6829560774012820a94b@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:57:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #487 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:57:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK626d735b Max-Forwards: 70 From: "Unknown" ;tag=as71663e69 To: Contact: Call-ID: 6b11e76002ce6829560774012820a94b@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:57:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK626d735b [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as71663e69 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 6b11e76002ce6829560774012820a94b@127.0.0.1:5060 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:57:32 GMT [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:57:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:57:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:57:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:57:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:57:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:57:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:57:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK626d735b;received=127.0.0.1 From: "Unknown" ;tag=as71663e69 To: ;tag=as71663e69 Call-ID: 6b11e76002ce6829560774012820a94b@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:57:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '6b11e76002ce6829560774012820a94b@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:57:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK626d735b;received=127.0.0.1 From: "Unknown" ;tag=as71663e69 To: ;tag=as71663e69 Call-ID: 6b11e76002ce6829560774012820a94b@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK626d735b;received=127.0.0.1 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as71663e69 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as71663e69 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 6b11e76002ce6829560774012820a94b@127.0.0.1:5060 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:57:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #487 [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '6b11e76002ce6829560774012820a94b@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:57:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 6b11e76002ce6829560774012820a94b@127.0.0.1:5060 [Aug 8 09:57:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '6b11e76002ce6829560774012820a94b@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:57:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:57:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:35] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:36] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:33600 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412656@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412656@127.0.0.1 [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:57:36] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:57:36] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:57:36] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412656@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:57:36] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:57:36] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:57:36] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:57:36] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:57:36] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:57:36] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:33600 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=33600 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as6a82306f Call-ID: 1344412656@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:57:36] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:33600 [Aug 8 09:57:36] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412656@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:57:38] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412626@127.0.0.1' [Aug 8 09:57:38] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412626@127.0.0.1 [Aug 8 09:57:38] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412626@127.0.0.1' Method: OPTIONS [Aug 8 09:57:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:40] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:57:45] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:46] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:46] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:49202 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412666@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412666@127.0.0.1 [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:57:46] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:57:46] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:57:46] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412666@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:57:46] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:57:46] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:57:46] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:57:46] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:57:46] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:57:46] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:49202 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=49202 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as48d1fbf7 Call-ID: 1344412666@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:57:46] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:49202 [Aug 8 09:57:46] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412666@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:57:48] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412636@127.0.0.1' [Aug 8 09:57:48] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412636@127.0.0.1 [Aug 8 09:57:48] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412636@127.0.0.1' Method: OPTIONS [Aug 8 09:57:50] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:51] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:57:55] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:56] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:57:56] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:37722 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412676@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412676@127.0.0.1 [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:57:56] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:57:56] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:57:56] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412676@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:57:56] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:57:56] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:57:56] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:57:56] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:57:56] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:57:56] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:37722 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=37722 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as43faf4e8 Call-ID: 1344412676@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:57:56] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:37722 [Aug 8 09:57:56] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412676@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:57:58] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412646@127.0.0.1' [Aug 8 09:57:58] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412646@127.0.0.1 [Aug 8 09:57:58] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412646@127.0.0.1' Method: OPTIONS [Aug 8 09:58:00] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:01] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:58:05] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:06] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:06] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:53731 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412686@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412686@127.0.0.1 [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:58:06] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:58:06] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:58:06] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412686@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:58:06] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:58:06] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:58:06] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:58:06] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:58:06] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:58:06] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:53731 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=53731 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as4e1640ef Call-ID: 1344412686@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:58:06] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:53731 [Aug 8 09:58:06] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412686@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:58:08] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412656@127.0.0.1' [Aug 8 09:58:08] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412656@127.0.0.1 [Aug 8 09:58:08] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412656@127.0.0.1' Method: OPTIONS [Aug 8 09:58:10] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:11] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:14] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:58:15] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:16] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:16] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:49223 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412696@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412696@127.0.0.1 [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:58:16] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:58:16] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:58:16] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412696@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:58:16] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:58:16] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:58:16] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:58:16] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:58:16] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:58:16] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:49223 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=49223 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as7b3efbec Call-ID: 1344412696@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:58:16] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:49223 [Aug 8 09:58:16] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412696@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:58:18] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412666@127.0.0.1' [Aug 8 09:58:18] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412666@127.0.0.1 [Aug 8 09:58:18] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412666@127.0.0.1' Method: OPTIONS [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> INVITE sip:2209@192.168.0.178:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-vqtvc78m0yau;rport From: ;tag=1del7f6fcr To: "Unknown" ;tag=as0fa7c2f6 Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;reg-id=1 X-Serialnumber: 0004132500A7 P-Key-Flags: keys="3" User-Agent: snom300/8.4.32 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 329 v=0 o=root 172176250 172176252 IN IP4 192.168.2.210 s=call c=IN IP4 192.168.2.210 t=0 0 m=audio 11986 RTP/SAVP 8 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:t/gyZ4pOluHvmshU2RCkno3w2gc93Ca/lz3lLjVQ a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendonly <-------------> [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 0 [ 42]: INVITE sip:2209@192.168.0.178:5060 SIP/2.0 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-vqtvc78m0yau;rport [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 2 [ 64]: From: ;tag=1del7f6fcr [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 3 [ 53]: To: "Unknown" ;tag=as0fa7c2f6 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 4 [ 60]: Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 7 [ 61]: Contact: ;reg-id=1 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 8 [ 28]: X-Serialnumber: 0004132500A7 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 9 [ 21]: P-Key-Flags: keys="3" [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 10 [ 26]: User-Agent: snom300/8.4.32 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 11 [ 23]: Accept: application/sdp [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 12 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 13 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 14 [ 40]: Supported: 100rel, replaces, from-change [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 16 [ 19]: Content-Length: 329 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 17 [ 0]: [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Body 0 [ 3]: v=0 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Body 1 [ 47]: o=root 172176250 172176252 IN IP4 192.168.2.210 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Body 2 [ 6]: s=call [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.2.210 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Body 5 [ 30]: m=audio 11986 RTP/SAVP 8 0 101 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:t/gyZ4pOluHvmshU2RCkno3w2gc93Ca/lz3lLjVQ [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Body 12 [ 10]: a=sendonly [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: --- (17 headers 13 lines) --- [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Aug 8 09:58:19] DEBUG[2371] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel, replaces, from-change" [Aug 8 09:58:19] DEBUG[2371] sip/reqresp_parser.c: Found SIP option: -100rel- [Aug 8 09:58:19] DEBUG[2371] sip/reqresp_parser.c: Matched SIP option: 100rel [Aug 8 09:58:19] DEBUG[2371] sip/reqresp_parser.c: Found SIP option: -replaces- [Aug 8 09:58:19] DEBUG[2371] sip/reqresp_parser.c: Matched SIP option: replaces [Aug 8 09:58:19] DEBUG[2371] sip/reqresp_parser.c: Found SIP option: -from-change- [Aug 8 09:58:19] DEBUG[2371] sip/reqresp_parser.c: Matched SIP option: from-change [Aug 8 09:58:19] DEBUG[2371] netsock2.c: Splitting '192.168.2.210:2048' into... [Aug 8 09:58:19] DEBUG[2371] netsock2.c: ...host '192.168.2.210' and port '2048'. [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: Sending to 192.168.2.210:2048 (NAT) [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Initializing initreq for method INVITE - callid 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Processing session-level SDP o=root 172176250 172176252 IN IP4 192.168.2.210... UNSUPPORTED OR FAILED. [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED OR FAILED. [Aug 8 09:58:19] DEBUG[2371] netsock2.c: Splitting '192.168.2.210' into... [Aug 8 09:58:19] DEBUG[2371] netsock2.c: ...host '192.168.2.210' and port ''. [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.2.210... OK. [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: Found RTP audio format 8 [Aug 8 09:58:19] DEBUG[2371] rtp_engine.c: Setting payload 8 based on m type on 0x41f0b5b0 [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: Found RTP audio format 0 [Aug 8 09:58:19] DEBUG[2371] rtp_engine.c: Setting payload 0 based on m type on 0x41f0b5b0 [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: Found RTP audio format 101 [Aug 8 09:58:19] DEBUG[2371] rtp_engine.c: Setting payload 101 based on m type on 0x41f0b5b0 [Aug 8 09:58:19] DEBUG[2371] res_srtp.c: Adding new policy for SSRC 797210039 [Aug 8 09:58:19] DEBUG[2371] sip/sdp_crypto.c: SRTP policy activated [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:t/gyZ4pOluHvmshU2RCkno3w2gc93Ca/lz3lLjVQ... OK. [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Aug 8 09:58:19] DEBUG[2371] rtp_engine.c: Incorporating payload 0 on 0x41f0b5b0 [Aug 8 09:58:19] DEBUG[2371] rtp_engine.c: Incorporating payload 8 on 0x41f0b5b0 [Aug 8 09:58:19] DEBUG[2371] rtp_engine.c: Incorporating payload 101 on 0x41f0b5b0 [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 8 09:58:19] DEBUG[2371] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c3bfc88' [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: Peer audio RTP is at port 192.168.2.210:11986 [Aug 8 09:58:19] DEBUG[2371] rtp_engine.c: Copying payload 0 from 0x41f0b5b0 to 0x1c3bfe50 [Aug 8 09:58:19] DEBUG[2371] rtp_engine.c: Copying payload 8 from 0x41f0b5b0 to 0x1c3bfe50 [Aug 8 09:58:19] DEBUG[2371] rtp_engine.c: Copying payload 101 from 0x41f0b5b0 to 0x1c3bfe50 [Aug 8 09:58:19] DEBUG[2371] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x1c3bfc88' [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: We have an owner, now see if we need to change this call [Aug 8 09:58:19] DEBUG[2371] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c3bfc88' [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Got a SIP re-invite for call 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: SIP/2210-00000001: This call is UP.... [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 192.168.2.210:2048 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-vqtvc78m0yau;received=192.168.2.210;rport=2048 From: ;tag=1del7f6fcr To: "Unknown" ;tag=as0fa7c2f6 Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 1 INVITE Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.2.210:2048 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Setting framing from config on incoming call [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: Audio is at 10866 [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: -- Done with adding codecs to SDP [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-vqtvc78m0yau;received=192.168.2.210;rport=2048 From: ;tag=1del7f6fcr To: "Unknown" ;tag=as0fa7c2f6 Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 1 INVITE Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 331 v=0 o=tvox 70257168 70257169 IN IP4 192.168.0.178 s=asterisk c=IN IP4 192.168.0.178 t=0 0 m=audio 10866 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:rrG/MleANMxuoCfjtPyp5qmmiZmZVJdBKyWgS1yP <------------> [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #496 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.2.210:2048 [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:58:19] VERBOSE[5008] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/2209-00000000 [Aug 8 09:58:19] DEBUG[5008] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:58:19] DEBUG[5008] channel.c: Got a FRAME_CONTROL (31) frame on channel SIP/2210-00000001 [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:58:19] DEBUG[5008] channel.c: Bridge stops bridging channels SIP/2209-00000000 and SIP/2210-00000001 [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:58:19] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2210 [Aug 8 09:58:19] DEBUG[2321] chan_sip.c: Checking device state for peer 2210 [Aug 8 09:58:19] DEBUG[2321] devicestate.c: Changing state for SIP/2210 - state 8 (On Hold) [Aug 8 09:58:19] DEBUG[2321] devicestate.c: device 'SIP/2210' state '8' [Aug 8 09:58:19] DEBUG[2980] manager.c: Examining event: Event: Hold Privilege: call,all Status: On Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 [Aug 8 09:58:19] DEBUG[2980] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/2209-00000000 UniqueID: 1344411528.0 Class: default [Aug 8 09:58:19] DEBUG[2980] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:19] DEBUG[2980] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:19] DEBUG[2984] manager.c: Examining event: Event: Hold Privilege: call,all Status: On Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 [Aug 8 09:58:19] DEBUG[2984] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/2209-00000000 UniqueID: 1344411528.0 Class: default [Aug 8 09:58:19] DEBUG[2984] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:19] DEBUG[2984] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:19] DEBUG[2988] manager.c: Examining event: Event: Hold Privilege: call,all Status: On Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 [Aug 8 09:58:19] DEBUG[2988] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/2209-00000000 UniqueID: 1344411528.0 Class: default [Aug 8 09:58:19] DEBUG[2988] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:19] DEBUG[2988] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:19] DEBUG[3176] manager.c: Examining event: Event: Hold Privilege: call,all Status: On Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 [Aug 8 09:58:19] DEBUG[3176] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/2209-00000000 UniqueID: 1344411528.0 Class: default [Aug 8 09:58:19] DEBUG[3176] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:19] DEBUG[3176] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:19] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPEER Value: SIP/2210-00000001 Uniqueid: 1344411528.0 [Aug 8 09:58:19] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPVTCALLID Value: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 Uniqueid: 1344411528.0 [Aug 8 09:58:19] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPEER Value: SIP/2209-00000000 Uniqueid: 1344411528.1 [Aug 8 09:58:19] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPVTCALLID Value: 3c39db8cc1a5-pv4xf5eh5w4v Uniqueid: 1344411528.1 [Aug 8 09:58:19] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPEER Value: SIP/2210-00000001 Uniqueid: 1344411528.0 [Aug 8 09:58:19] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPVTCALLID Value: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 Uniqueid: 1344411528.0 [Aug 8 09:58:19] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPEER Value: SIP/2209-00000000 Uniqueid: 1344411528.1 [Aug 8 09:58:19] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPVTCALLID Value: 3c39db8cc1a5-pv4xf5eh5w4v Uniqueid: 1344411528.1 [Aug 8 09:58:19] DEBUG[2323] app_queue.c: Extension '2210@telenia_localextensions' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. [Aug 8 09:58:19] DEBUG[2394] app_queue.c: Device 'SIP/2210' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[2980] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 16 [Aug 8 09:58:19] DEBUG[2984] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 16 [Aug 8 09:58:19] DEBUG[2988] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 16 [Aug 8 09:58:19] DEBUG[3176] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 16 [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> ACK sip:2209@192.168.0.178:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-3edaswxjkosj;rport From: ;tag=1del7f6fcr To: "Unknown" ;tag=as0fa7c2f6 Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 1 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 0 [ 39]: ACK sip:2209@192.168.0.178:5060 SIP/2.0 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-3edaswxjkosj;rport [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 2 [ 64]: From: ;tag=1del7f6fcr [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 3 [ 53]: To: "Unknown" ;tag=as0fa7c2f6 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 4 [ 60]: Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 7 [ 61]: Contact: ;reg-id=1 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Header 9 [ 0]: [Aug 8 09:58:19] VERBOSE[2371] chan_sip.c: --- (9 headers 0 lines) --- [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #496 [Aug 8 09:58:19] DEBUG[2371] chan_sip.c: Stopping retransmission on '690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060' of Response 1: Match Found [Aug 8 09:58:19] DEBUG[5008] channel.c: Set channel SIP/2209-00000000 to write format slin [Aug 8 09:58:19] DEBUG[5008] res_musiconhold.c: SIP/2209-00000000 Opened file 0 '/var/lib/telenia/php/siti/t-vox/sounds/ivr/03_tchaikovsky-nocturn//03_tchaikovsky-nocturn' [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: Difference is 4160, ms is 540 [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:19] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:20] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: RTCP NAT: Got RTCP from other end. Now sending to address 192.168.2.210:11987 [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 24 bytes [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:21] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:22] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:23] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:24] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: No remote address on RTP instance '0x1c3bfc88' so dropping frame [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> INVITE sip:2209@192.168.0.178:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-kob82pie5gip;rport From: ;tag=1del7f6fcr To: "Unknown" ;tag=as0fa7c2f6 Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 2 INVITE Max-Forwards: 70 Contact: ;reg-id=1 X-Serialnumber: 0004132500A7 P-Key-Flags: keys="3" User-Agent: snom300/8.4.32 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 329 v=0 o=root 172176250 172176253 IN IP4 192.168.2.210 s=call c=IN IP4 192.168.2.210 t=0 0 m=audio 11986 RTP/SAVP 8 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:t/gyZ4pOluHvmshU2RCkno3w2gc93Ca/lz3lLjVQ a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 0 [ 42]: INVITE sip:2209@192.168.0.178:5060 SIP/2.0 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-kob82pie5gip;rport [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 2 [ 64]: From: ;tag=1del7f6fcr [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 3 [ 53]: To: "Unknown" ;tag=as0fa7c2f6 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 4 [ 60]: Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 5 [ 14]: CSeq: 2 INVITE [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 7 [ 61]: Contact: ;reg-id=1 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 8 [ 28]: X-Serialnumber: 0004132500A7 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 9 [ 21]: P-Key-Flags: keys="3" [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 10 [ 26]: User-Agent: snom300/8.4.32 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 11 [ 23]: Accept: application/sdp [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 12 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 13 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 14 [ 40]: Supported: 100rel, replaces, from-change [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 16 [ 19]: Content-Length: 329 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 17 [ 0]: [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Body 0 [ 3]: v=0 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Body 1 [ 47]: o=root 172176250 172176253 IN IP4 192.168.2.210 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Body 2 [ 6]: s=call [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.2.210 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Body 5 [ 30]: m=audio 11986 RTP/SAVP 8 0 101 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:t/gyZ4pOluHvmshU2RCkno3w2gc93Ca/lz3lLjVQ [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Body 12 [ 10]: a=sendrecv [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: --- (17 headers 13 lines) --- [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Aug 8 09:58:24] DEBUG[2371] netsock2.c: Splitting '192.168.2.210:2048' into... [Aug 8 09:58:24] DEBUG[2371] netsock2.c: ...host '192.168.2.210' and port '2048'. [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: Sending to 192.168.2.210:2048 (NAT) [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Initializing initreq for method INVITE - callid 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Processing session-level SDP o=root 172176250 172176253 IN IP4 192.168.2.210... UNSUPPORTED OR FAILED. [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED OR FAILED. [Aug 8 09:58:24] DEBUG[2371] netsock2.c: Splitting '192.168.2.210' into... [Aug 8 09:58:24] DEBUG[2371] netsock2.c: ...host '192.168.2.210' and port ''. [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.2.210... OK. [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: Found RTP audio format 8 [Aug 8 09:58:24] DEBUG[2371] rtp_engine.c: Setting payload 8 based on m type on 0x41f0b5b0 [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: Found RTP audio format 0 [Aug 8 09:58:24] DEBUG[2371] rtp_engine.c: Setting payload 0 based on m type on 0x41f0b5b0 [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: Found RTP audio format 101 [Aug 8 09:58:24] DEBUG[2371] rtp_engine.c: Setting payload 101 based on m type on 0x41f0b5b0 [Aug 8 09:58:24] DEBUG[2371] res_srtp.c: Adding new policy for SSRC 797210039 [Aug 8 09:58:24] DEBUG[2371] sip/sdp_crypto.c: SRTP policy activated [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:t/gyZ4pOluHvmshU2RCkno3w2gc93Ca/lz3lLjVQ... OK. [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 8 09:58:24] DEBUG[2371] rtp_engine.c: Incorporating payload 0 on 0x41f0b5b0 [Aug 8 09:58:24] DEBUG[2371] rtp_engine.c: Incorporating payload 8 on 0x41f0b5b0 [Aug 8 09:58:24] DEBUG[2371] rtp_engine.c: Incorporating payload 101 on 0x41f0b5b0 [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 8 09:58:24] DEBUG[2371] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c3bfc88' [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: Peer audio RTP is at port 192.168.2.210:11986 [Aug 8 09:58:24] DEBUG[2371] rtp_engine.c: Copying payload 0 from 0x41f0b5b0 to 0x1c3bfe50 [Aug 8 09:58:24] DEBUG[2371] rtp_engine.c: Copying payload 8 from 0x41f0b5b0 to 0x1c3bfe50 [Aug 8 09:58:24] DEBUG[2371] rtp_engine.c: Copying payload 101 from 0x41f0b5b0 to 0x1c3bfe50 [Aug 8 09:58:24] DEBUG[2371] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x1c3bfc88' [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: We have an owner, now see if we need to change this call [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Got a SIP re-invite for call 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: SIP/2210-00000001: This call is UP.... [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 192.168.2.210:2048 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-kob82pie5gip;received=192.168.2.210;rport=2048 From: ;tag=1del7f6fcr To: "Unknown" ;tag=as0fa7c2f6 Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 2 INVITE Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.2.210:2048 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Setting framing from config on incoming call [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: Audio is at 10866 [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: -- Done with adding codecs to SDP [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-kob82pie5gip;received=192.168.2.210;rport=2048 From: ;tag=1del7f6fcr To: "Unknown" ;tag=as0fa7c2f6 Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 2 INVITE Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 331 v=0 o=tvox 70257168 70257170 IN IP4 192.168.0.178 s=asterisk c=IN IP4 192.168.0.178 t=0 0 m=audio 10866 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:rrG/MleANMxuoCfjtPyp5qmmiZmZVJdBKyWgS1yP <------------> [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #497 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.2.210:2048 [Aug 8 09:58:24] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2210 [Aug 8 09:58:24] DEBUG[2321] chan_sip.c: Checking device state for peer 2210 [Aug 8 09:58:24] DEBUG[2321] devicestate.c: Changing state for SIP/2210 - state 2 (In use) [Aug 8 09:58:24] DEBUG[2321] devicestate.c: device 'SIP/2210' state '2' [Aug 8 09:58:24] DEBUG[2980] manager.c: Examining event: Event: Hold Privilege: call,all Status: Off Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 [Aug 8 09:58:24] DEBUG[2984] manager.c: Examining event: Event: Hold Privilege: call,all Status: Off Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 [Aug 8 09:58:24] DEBUG[2988] manager.c: Examining event: Event: Hold Privilege: call,all Status: Off Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 [Aug 8 09:58:24] DEBUG[3176] manager.c: Examining event: Event: Hold Privilege: call,all Status: Off Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:58:24] DEBUG[2323] app_queue.c: Extension '2210@telenia_localextensions' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 8 09:58:24] DEBUG[2394] app_queue.c: Device 'SIP/2210' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 8 09:58:24] DEBUG[2980] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 1 [Aug 8 09:58:24] DEBUG[2984] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 1 [Aug 8 09:58:24] DEBUG[2988] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 1 [Aug 8 09:58:24] DEBUG[3176] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 1 [Aug 8 09:58:24] VERBOSE[5008] res_musiconhold.c: -- Stopped music on hold on SIP/2209-00000000 [Aug 8 09:58:24] DEBUG[5008] channel.c: Set channel SIP/2209-00000000 to write format alaw [Aug 8 09:58:24] DEBUG[5008] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 8 09:58:24] DEBUG[2980] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/2209-00000000 UniqueID: 1344411528.0 [Aug 8 09:58:24] DEBUG[2984] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/2209-00000000 UniqueID: 1344411528.0 [Aug 8 09:58:24] DEBUG[2988] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/2209-00000000 UniqueID: 1344411528.0 [Aug 8 09:58:24] DEBUG[3176] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/2209-00000000 UniqueID: 1344411528.0 [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x1c3bfc88' [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:58:24] DEBUG[5008] channel.c: Got a FRAME_CONTROL (31) frame on channel SIP/2210-00000001 [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:58:24] DEBUG[2980] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:24] DEBUG[2984] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:24] DEBUG[2988] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:24] DEBUG[3176] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:24] DEBUG[5008] channel.c: Bridge stops bridging channels SIP/2209-00000000 and SIP/2210-00000001 [Aug 8 09:58:24] DEBUG[2980] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:24] DEBUG[2984] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:24] DEBUG[2988] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:24] DEBUG[3176] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:58:24] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPEER Value: SIP/2210-00000001 Uniqueid: 1344411528.0 [Aug 8 09:58:24] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPVTCALLID Value: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 Uniqueid: 1344411528.0 [Aug 8 09:58:24] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPEER Value: SIP/2209-00000000 Uniqueid: 1344411528.1 [Aug 8 09:58:24] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPVTCALLID Value: 3c39db8cc1a5-pv4xf5eh5w4v Uniqueid: 1344411528.1 [Aug 8 09:58:24] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPEER Value: SIP/2210-00000001 Uniqueid: 1344411528.0 [Aug 8 09:58:24] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: BRIDGEPVTCALLID Value: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 Uniqueid: 1344411528.0 [Aug 8 09:58:24] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPEER Value: SIP/2209-00000000 Uniqueid: 1344411528.1 [Aug 8 09:58:24] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: BRIDGEPVTCALLID Value: 3c39db8cc1a5-pv4xf5eh5w4v Uniqueid: 1344411528.1 [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> ACK sip:2209@192.168.0.178:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-oas7vayffbyv;rport From: ;tag=1del7f6fcr To: "Unknown" ;tag=as0fa7c2f6 Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 2 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 0 [ 39]: ACK sip:2209@192.168.0.178:5060 SIP/2.0 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.2.210:2048;branch=z9hG4bK-oas7vayffbyv;rport [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 2 [ 64]: From: ;tag=1del7f6fcr [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 3 [ 53]: To: "Unknown" ;tag=as0fa7c2f6 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 4 [ 60]: Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 5 [ 11]: CSeq: 2 ACK [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 7 [ 61]: Contact: ;reg-id=1 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Header 9 [ 0]: [Aug 8 09:58:24] VERBOSE[2371] chan_sip.c: --- (9 headers 0 lines) --- [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #497 [Aug 8 09:58:24] DEBUG[2371] chan_sip.c: Stopping retransmission on '690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060' of Response 2: Match Found [Aug 8 09:58:24] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:24] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 10 [Aug 8 09:58:26] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:26] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:41157 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412706@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412706@127.0.0.1 [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:58:26] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:58:26] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:58:26] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412706@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:58:26] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:58:26] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:58:26] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:58:26] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:58:26] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:58:26] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:41157 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=41157 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as51276ba8 Call-ID: 1344412706@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:58:26] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:41157 [Aug 8 09:58:26] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412706@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:58:26] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:28] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412676@127.0.0.1' [Aug 8 09:58:28] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412676@127.0.0.1 [Aug 8 09:58:28] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412676@127.0.0.1' Method: OPTIONS [Aug 8 09:58:28] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:29] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:30] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:31] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 4bcec4be5e164a4b36ed09db20e5402e@(null) - OPTIONS (No RTP) [Aug 8 09:58:32] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:58:32] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Initializing initreq for method OPTIONS - callid 14faabe04825c49b070eb33054e2953d@127.0.0.1:5060 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK497e6760 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as0497a394 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 14faabe04825c49b070eb33054e2953d@127.0.0.1:5060 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:58:32 GMT [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:58:32] VERBOSE[2371] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK497e6760 Max-Forwards: 70 From: "Unknown" ;tag=as0497a394 To: Contact: Call-ID: 14faabe04825c49b070eb33054e2953d@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:58:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #500 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:58:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK497e6760 Max-Forwards: 70 From: "Unknown" ;tag=as0497a394 To: Contact: Call-ID: 14faabe04825c49b070eb33054e2953d@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: asterisk Date: Wed, 08 Aug 2012 07:58:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 0 [ 29]: OPTIONS sip:127.0.0.1 SIP/2.0 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 1 [ 54]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK497e6760 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 3 [ 54]: From: "Unknown" ;tag=as0497a394 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 4 [ 19]: To: [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 5 [ 37]: Contact: [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 6 [ 56]: Call-ID: 14faabe04825c49b070eb33054e2953d@127.0.0.1:5060 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 8 [ 20]: User-Agent: asterisk [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 9 [ 35]: Date: Wed, 08 Aug 2012 07:58:32 GMT [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 13 [ 0]: [Aug 8 09:58:32] VERBOSE[2371] chan_sip.c: --- (13 headers 0 lines) --- [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:58:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:58:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:58:32] DEBUG[2371] netsock2.c: Splitting '127.0.0.1' into... [Aug 8 09:58:32] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:58:32] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:58:32] VERBOSE[2371] chan_sip.c: <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK497e6760;received=127.0.0.1 From: "Unknown" ;tag=as0497a394 To: ;tag=as0497a394 Call-ID: 14faabe04825c49b070eb33054e2953d@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:5060 [Aug 8 09:58:32] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '14faabe04825c49b070eb33054e2953d@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) [Aug 8 09:58:32] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK497e6760;received=127.0.0.1 From: "Unknown" ;tag=as0497a394 To: ;tag=as0497a394 Call-ID: 14faabe04825c49b070eb33054e2953d@127.0.0.1:5060 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK497e6760;received=127.0.0.1 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 2 [ 54]: From: "Unknown" ;tag=as0497a394 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 3 [ 34]: To: ;tag=as0497a394 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 4 [ 56]: Call-ID: 14faabe04825c49b070eb33054e2953d@127.0.0.1:5060 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Server: asterisk [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 9 [ 37]: Contact: [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:58:32] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #500 [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Stopping retransmission on '14faabe04825c49b070eb33054e2953d@127.0.0.1:5060' of Request 102: Match Found [Aug 8 09:58:32] DEBUG[2371] chan_sip.c: Destroying SIP dialog 14faabe04825c49b070eb33054e2953d@127.0.0.1:5060 [Aug 8 09:58:32] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '14faabe04825c49b070eb33054e2953d@127.0.0.1:5060' Method: OPTIONS [Aug 8 09:58:32] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:34] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:58:34] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:34] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:36] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:36] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:56391 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412716@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412716@127.0.0.1 [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:58:36] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:58:36] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:58:36] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412716@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:58:36] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:58:36] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:58:36] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:58:36] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:58:36] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:58:36] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:56391 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=56391 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as628d9f7f Call-ID: 1344412716@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:58:36] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:56391 [Aug 8 09:58:36] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412716@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:58:36] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:38] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412686@127.0.0.1' [Aug 8 09:58:38] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412686@127.0.0.1 [Aug 8 09:58:38] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412686@127.0.0.1' Method: OPTIONS [Aug 8 09:58:38] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:39] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:40] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:41] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:42] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:44] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:58:44] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:44] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:46] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:46] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:51570 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412726@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412726@127.0.0.1 [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:58:46] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:58:46] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:58:46] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412726@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:58:46] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:58:46] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:58:46] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:58:46] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:58:46] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:58:46] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:51570 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=51570 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as2667d0b0 Call-ID: 1344412726@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:58:46] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:51570 [Aug 8 09:58:46] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412726@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:58:46] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:48] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412696@127.0.0.1' [Aug 8 09:58:48] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412696@127.0.0.1 [Aug 8 09:58:48] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412696@127.0.0.1' Method: OPTIONS [Aug 8 09:58:48] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:49] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:50] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:51] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:52] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:54] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:58:54] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:54] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:56] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:58:56] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:44957 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412736@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412736@127.0.0.1 [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:58:56] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:58:56] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:58:56] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412736@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:58:56] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:58:56] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:58:56] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:58:56] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:58:56] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:58:56] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:44957 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=44957 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as4c929258 Call-ID: 1344412736@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:58:56] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:44957 [Aug 8 09:58:56] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412736@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:58:56] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:58] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412706@127.0.0.1' [Aug 8 09:58:58] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412706@127.0.0.1 [Aug 8 09:58:58] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412706@127.0.0.1' Method: OPTIONS [Aug 8 09:58:58] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:58:59] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:59:00] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:59:01] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:59:02] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:59:04] DEBUG[2982] manager.c: Running action 'Command' [Aug 8 09:59:04] DEBUG[5008] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 8 09:59:04] WARNING[5008] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Aug 8 09:59:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.1.102:2052 ---> BYE sip:2210@192.168.0.178:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-9zdipd0rvsa5;rport From: "2209" ;tag=gbdfdvpkud To: ;tag=as0bbc9cb2 Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v CSeq: 3 BYE Max-Forwards: 70 Contact: ;reg-id=1 User-Agent: snom300/8.4.32 RTP-RxStat: Total_Rx_Pkts=58461,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=5099182 RTP-TxStat: Total_Tx_Pkts=60529,Tx_Pkts=60529,Remote_Tx_Pkts=-782451174 Content-Length: 0 <-------------> [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 0 [ 39]: BYE sip:2210@192.168.0.178:5060 SIP/2.0 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-9zdipd0rvsa5;rport [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 2 [ 52]: From: "2209" ;tag=gbdfdvpkud [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 3 [ 43]: To: ;tag=as0bbc9cb2 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 4 [ 34]: Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 5 [ 11]: CSeq: 3 BYE [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 7 [ 61]: Contact: ;reg-id=1 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 8 [ 26]: User-Agent: snom300/8.4.32 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 9 [ 84]: RTP-RxStat: Total_Rx_Pkts=58461,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=5099182 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 10 [ 71]: RTP-TxStat: Total_Tx_Pkts=60529,Tx_Pkts=60529,Remote_Tx_Pkts=-782451174 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:59:05] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Initializing initreq for method BYE - callid 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:59:05] DEBUG[2371] netsock2.c: Splitting '192.168.1.102:2052' into... [Aug 8 09:59:05] DEBUG[2371] netsock2.c: ...host '192.168.1.102' and port '2052'. [Aug 8 09:59:05] VERBOSE[2371] chan_sip.c: Sending to 192.168.1.102:2052 (NAT) [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: RTPAUDIOQOS Value: ssrc=1420768450;themssrc=3217693234;lp=0;rxjitter=0.003904;rxcount=60520;txjitter=0.000000;txcount=58470;rlp=1;rtt=59604.391000 Uniqueid: 1344411528.0 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: RTPAUDIOQOSBRIDGED Value: ssrc=1420768450;themssrc=3217693234;lp=0;rxjitter=0.003904;rxcount=60520;txjitter=0.000000;txcount=58470;rlp=1;rtt=59604.391000 Uniqueid: 1344411528.1 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1344411528.0 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: RTPAUDIOQOSJITTERBRIDGED Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1344411528.1 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1344411528.0 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: RTPAUDIOQOSLOSSBRIDGED Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1344411528.1 [Aug 8 09:59:05] DEBUG[2371] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c3a7dc8' [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Session timer stopped: 288 - 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:59:05] DEBUG[5008] channel.c: Didn't get a frame from channel: SIP/2209-00000000 [Aug 8 09:59:05] DEBUG[5008] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 8 09:59:05] DEBUG[5008] channel.c: Bridge stops bridging channels SIP/2209-00000000 and SIP/2210-00000001 [Aug 8 09:59:05] DEBUG[5008] cdr.c: Dropping CDR ! [Aug 8 09:59:05] DEBUG[5008] channel.c: Hanging up channel 'SIP/2210-00000001' [Aug 8 09:59:05] DEBUG[2980] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:59:05] DEBUG[2984] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:59:05] DEBUG[2988] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:59:05] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '3c39db8cc1a5-pv4xf5eh5w4v' in 32000 ms (Method: BYE) [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Received bye, issuing owner hangup [Aug 8 09:59:05] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.102:2052 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102:2052;branch=z9hG4bK-9zdipd0rvsa5;received=192.168.1.102;rport=2052 From: "2209" ;tag=gbdfdvpkud To: ;tag=as0bbc9cb2 Call-ID: 3c39db8cc1a5-pv4xf5eh5w4v CSeq: 3 BYE Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.102:2052 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1344411528.0 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1344411528.1 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: RTPAUDIOQOS Value: ssrc=797210039;themssrc=3313322812;lp=0;rxjitter=0.003662;rxcount=58219;txjitter=0.000000;txcount=60269;rlp=1;rtt=0.004000 Uniqueid: 1344411528.1 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: RTPAUDIOQOSBRIDGED Value: ssrc=797210039;themssrc=3313322812;lp=0;rxjitter=0.003662;rxcount=58219;txjitter=0.000000;txcount=60269;rlp=1;rtt=0.004000 Uniqueid: 1344411528.0 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1344411528.1 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: RTPAUDIOQOSJITTERBRIDGED Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1344411528.0 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1344411528.1 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: RTPAUDIOQOSLOSSBRIDGED Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1344411528.0 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1344411528.1 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1344411528.0 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2209-00000000 Channel2: SIP/2210-00000001 Uniqueid1: 1344411528.0 Uniqueid2: 1344411528.1 CallerID1: 2209 CallerID2: 2210 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: ANSWEREDTIME Value: 1211 Uniqueid: 1344411528.0 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: DIALEDTIME Value: 1217 Uniqueid: 1344411528.0 [Aug 8 09:59:05] DEBUG[5008] chan_sip.c: Hangup call SIP/2210-00000001, SIP callid 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:59:05] DEBUG[5008] chan_sip.c: update_call_counter(2210) - decrement call limit counter on hangup [Aug 8 09:59:05] DEBUG[5008] chan_sip.c: Updating call counter for outgoing call [Aug 8 09:59:05] DEBUG[5008] chan_sip.c: Call to peer '2210' removed from call limit 2147483647 [Aug 8 09:59:05] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2210 [Aug 8 09:59:05] DEBUG[2321] chan_sip.c: Checking device state for peer 2210 [Aug 8 09:59:05] DEBUG[2321] devicestate.c: Changing state for SIP/2210 - state 1 (Not in use) [Aug 8 09:59:05] DEBUG[2321] devicestate.c: device 'SIP/2210' state '1' [Aug 8 09:59:05] DEBUG[2323] app_queue.c: Extension '2210@telenia_localextensions' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 8 09:59:05] DEBUG[2394] app_queue.c: Device 'SIP/2210' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 8 09:59:05] DEBUG[2980] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 0 [Aug 8 09:59:05] DEBUG[2984] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 0 [Aug 8 09:59:05] DEBUG[2988] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 0 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2210 Context: telenia_localextensions Hint: SIP/2210 Status: 0 [Aug 8 09:59:05] DEBUG[5008] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c3bfc88' [Aug 8 09:59:05] VERBOSE[5008] chan_sip.c: Scheduling destruction of SIP dialog '690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060' in 32000 ms (Method: ACK) [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: RTPAUDIOQOS Value: ssrc=797210039;themssrc=3313322812;lp=0;rxjitter=0.003662;rxcount=58219;txjitter=0.000000;txcount=60269;rlp=1;rtt=0.004000 Uniqueid: 1344411528.1 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1344411528.1 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1344411528.1 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1344411528.1 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2210-00000001 Variable: RTPAUDIOQOS Value: ssrc=797210039;themssrc=3313322812;lp=0;rxjitter=0.003662;rxcount=58219;txjitter=0.000000;txcount=60269;rlp=1;rtt=0.004000 Uniqueid: 1344411528.1 [Aug 8 09:59:05] DEBUG[5008] chan_sip.c: Strict routing enforced for session 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:59:05] VERBOSE[5008] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 8 09:59:05] DEBUG[5008] netsock2.c: Splitting '192.168.2.210:2048' into... [Aug 8 09:59:05] DEBUG[5008] netsock2.c: ...host '192.168.2.210' and port '2048'. [Aug 8 09:59:05] VERBOSE[5008] chan_sip.c: set_destination: set destination to 192.168.2.210:2048 [Aug 8 09:59:05] VERBOSE[5008] chan_sip.c: Reliably Transmitting (NAT) to 192.168.2.210:2048: BYE sip:2210@192.168.2.210:2048;line=nh5ckpq1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK7d7fa07e;rport Max-Forwards: 70 From: "Unknown" ;tag=as0fa7c2f6 To: ;tag=1del7f6fcr Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 103 BYE User-Agent: asterisk X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 8 09:59:05] DEBUG[5008] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #509 [Aug 8 09:59:05] DEBUG[5008] chan_sip.c: Trying to put 'BYE sip:221' onto UDP socket destined for 192.168.2.210:2048 [Aug 8 09:59:05] DEBUG[2980] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 CallerIDNum: 2210 CallerIDName: ConnectedLineNum: 2209 ConnectedLineName: Unknown Cause: 16 Cause-txt: Normal Clearing [Aug 8 09:59:05] DEBUG[2984] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 CallerIDNum: 2210 CallerIDName: ConnectedLineNum: 2209 ConnectedLineName: Unknown Cause: 16 Cause-txt: Normal Clearing [Aug 8 09:59:05] DEBUG[2988] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 CallerIDNum: 2210 CallerIDName: ConnectedLineNum: 2209 ConnectedLineName: Unknown Cause: 16 Cause-txt: Normal Clearing [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/2210-00000001 Uniqueid: 1344411528.1 CallerIDNum: 2210 CallerIDName: ConnectedLineNum: 2209 ConnectedLineName: Unknown Cause: 16 Cause-txt: Normal Clearing [Aug 8 09:59:05] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2210 [Aug 8 09:59:05] DEBUG[2321] chan_sip.c: Checking device state for peer 2210 [Aug 8 09:59:05] DEBUG[2321] devicestate.c: Changing state for SIP/2210 - state 1 (Not in use) [Aug 8 09:59:05] DEBUG[2321] devicestate.c: device 'SIP/2210' state '1' [Aug 8 09:59:05] DEBUG[2394] app_queue.c: Device 'SIP/2210' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/2209-00000000 Variable: DIALSTATUS Value: ANSWER Uniqueid: 1344411528.0 [Aug 8 09:59:05] DEBUG[2980] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/2209-00000000 UniqueID: 1344411528.0 DialStatus: ANSWER [Aug 8 09:59:05] DEBUG[2984] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/2209-00000000 UniqueID: 1344411528.0 DialStatus: ANSWER [Aug 8 09:59:05] DEBUG[2988] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/2209-00000000 UniqueID: 1344411528.0 DialStatus: ANSWER [Aug 8 09:59:05] DEBUG[5008] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/2209-00000000 UniqueID: 1344411528.0 DialStatus: ANSWER [Aug 8 09:59:05] DEBUG[5008] pbx.c: Spawn extension (test_issue,2210,1) exited non-zero on 'SIP/2209-00000000' [Aug 8 09:59:05] VERBOSE[5008] pbx.c: == Spawn extension (test_issue, 2210, 1) exited non-zero on 'SIP/2209-00000000' [Aug 8 09:59:05] DEBUG[5008] channel.c: Soft-Hanging up channel 'SIP/2209-00000000' [Aug 8 09:59:05] DEBUG[5008] channel.c: Hanging up channel 'SIP/2209-00000000' [Aug 8 09:59:05] DEBUG[5008] chan_sip.c: Hangup call SIP/2209-00000000, SIP callid 3c39db8cc1a5-pv4xf5eh5w4v [Aug 8 09:59:05] DEBUG[5008] chan_sip.c: update_call_counter(2209) - decrement call limit counter on hangup [Aug 8 09:59:05] DEBUG[5008] chan_sip.c: Updating call counter for incoming call [Aug 8 09:59:05] DEBUG[5008] chan_sip.c: Call from peer '2209' removed from call limit 2147483647 [Aug 8 09:59:05] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2209 [Aug 8 09:59:05] DEBUG[2321] chan_sip.c: Checking device state for peer 2209 [Aug 8 09:59:05] DEBUG[2321] devicestate.c: Changing state for SIP/2209 - state 1 (Not in use) [Aug 8 09:59:05] DEBUG[2321] devicestate.c: device 'SIP/2209' state '1' [Aug 8 09:59:05] DEBUG[2323] app_queue.c: Extension '2209@telenia_localextensions' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 8 09:59:05] DEBUG[2394] app_queue.c: Device 'SIP/2209' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 8 09:59:05] DEBUG[2980] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2209 Context: telenia_localextensions Hint: SIP/2209 Status: 0 [Aug 8 09:59:05] DEBUG[2984] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2209 Context: telenia_localextensions Hint: SIP/2209 Status: 0 [Aug 8 09:59:05] DEBUG[2988] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2209 Context: telenia_localextensions Hint: SIP/2209 Status: 0 [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2209 Context: telenia_localextensions Hint: SIP/2209 Status: 0 [Aug 8 09:59:05] DEBUG[5008] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c3a7dc8' [Aug 8 09:59:05] DEBUG[2980] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/2209-00000000 Uniqueid: 1344411528.0 CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: Unknown Cause: 16 Cause-txt: Normal Clearing [Aug 8 09:59:05] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:192.168.2.210:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK7d7fa07e;rport=5060 From: "Unknown" ;tag=as0fa7c2f6 To: ;tag=1del7f6fcr Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 CSeq: 103 BYE Contact: ;reg-id=1 User-Agent: snom300/8.4.32 RTP-RxStat: Total_Rx_Pkts=60259,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=60270,Tx_Pkts=2059,Remote_Tx_Pkts=60213 Content-Length: 0 <-------------> [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.0.178:5060;branch=z9hG4bK7d7fa07e;rport=5060 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 2 [ 55]: From: "Unknown" ;tag=as0fa7c2f6 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 3 [ 62]: To: ;tag=1del7f6fcr [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 4 [ 60]: Call-ID: 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 6 [ 61]: Contact: ;reg-id=1 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 7 [ 26]: User-Agent: snom300/8.4.32 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 8 [ 78]: RTP-RxStat: Total_Rx_Pkts=60259,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 9 [ 65]: RTP-TxStat: Total_Tx_Pkts=60270,Tx_Pkts=2059,Remote_Tx_Pkts=60213 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Header 11 [ 0]: [Aug 8 09:59:05] VERBOSE[2371] chan_sip.c: --- (11 headers 0 lines) --- [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #509 [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Stopping retransmission on '690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060' of Request 103: Match Found [Aug 8 09:59:05] VERBOSE[2371] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [Aug 8 09:59:05] DEBUG[2371] chan_sip.c: Destroying SIP dialog 690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060 [Aug 8 09:59:05] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '690997c26e07339706d5f8850f46c9b6@192.168.0.178:5060' Method: ACK [Aug 8 09:59:05] DEBUG[2371] rtp_engine.c: Destroyed RTP instance '0x1c3bfc88' [Aug 8 09:59:05] DEBUG[2984] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/2209-00000000 Uniqueid: 1344411528.0 CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: Unknown Cause: 16 Cause-txt: Normal Clearing [Aug 8 09:59:05] DEBUG[2988] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/2209-00000000 Uniqueid: 1344411528.0 CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: Unknown Cause: 16 Cause-txt: Normal Clearing [Aug 8 09:59:05] DEBUG[3176] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/2209-00000000 Uniqueid: 1344411528.0 CallerIDNum: 2209 CallerIDName: Unknown ConnectedLineNum: ConnectedLineName: Unknown Cause: 16 Cause-txt: Normal Clearing [Aug 8 09:59:05] DEBUG[2321] devicestate.c: No provider found, checking channel drivers for SIP - 2209 [Aug 8 09:59:05] DEBUG[2321] chan_sip.c: Checking device state for peer 2209 [Aug 8 09:59:05] DEBUG[2321] devicestate.c: Changing state for SIP/2209 - state 1 (Not in use) [Aug 8 09:59:05] DEBUG[2321] devicestate.c: device 'SIP/2209' state '1' [Aug 8 09:59:05] DEBUG[2394] app_queue.c: Device 'SIP/2209' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 8 09:59:06] VERBOSE[2371] chan_sip.c: <--- SIP read from UDP:127.0.0.1:49417 ---> OPTIONS sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:9999;rport From: "TVOX Check" ;tag=as7a91ea1e To: Contact: Call-ID: 1344412746@127.0.0.1 CSeq: 102 OPTIONS User-Agent: Test TVOX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Expires: 1 <-------------> [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Header 0 [ 34]: OPTIONS sip:127.0.0.1:5060 SIP/2.0 [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Header 1 [ 37]: Via: SIP/2.0/UDP 127.0.0.1:9999;rport [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Header 2 [ 62]: From: "TVOX Check" ;tag=as7a91ea1e [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Header 3 [ 35]: To: [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Header 4 [ 37]: Contact: [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Header 5 [ 29]: Call-ID: 1344412746@127.0.0.1 [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Header 7 [ 21]: User-Agent: Test TVOX [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Header 11 [ 10]: Expires: 1 [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Header 12 [ 0]: [Aug 8 09:59:06] VERBOSE[2371] chan_sip.c: --- (12 headers 0 lines) --- [Aug 8 09:59:06] DEBUG[2371] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Aug 8 09:59:06] DEBUG[2371] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 127.0.0.1:5060 [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Allocating new SIP dialog for 1344412746@127.0.0.1 - OPTIONS (No RTP) [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Aug 8 09:59:06] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:5060' into... [Aug 8 09:59:06] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:59:06] DEBUG[2371] netsock2.c: Splitting '127.0.0.1:9999' into... [Aug 8 09:59:06] DEBUG[2371] netsock2.c: ...host '127.0.0.1' and port ''. [Aug 8 09:59:06] VERBOSE[2371] chan_sip.c: Looking for s in telenia_inbound_route_internal (domain 127.0.0.1) [Aug 8 09:59:06] VERBOSE[2371] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:49417 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:9999;received=127.0.0.1;rport=49417 From: "TVOX Check" ;tag=as7a91ea1e To: ;tag=as52d13228 Call-ID: 1344412746@127.0.0.1 CSeq: 102 OPTIONS Server: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Aug 8 09:59:06] DEBUG[2371] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 127.0.0.1:49417 [Aug 8 09:59:06] VERBOSE[2371] chan_sip.c: Scheduling destruction of SIP dialog '1344412746@127.0.0.1' in 32000 ms (Method: OPTIONS) [Aug 8 09:59:08] DEBUG[2371] chan_sip.c: Auto destroying SIP dialog '1344412716@127.0.0.1' [Aug 8 09:59:08] DEBUG[2371] chan_sip.c: Destroying SIP dialog 1344412716@127.0.0.1 [Aug 8 09:59:08] VERBOSE[2371] chan_sip.c: Really destroying SIP dialog '1344412716@127.0.0.1' Method: OPTIONS