[Jan 17 21:32:19] VERBOSE[3258] config.c: == Parsing '/etc/asterisk/logger.conf': [Jan 17 21:32:19] DEBUG[3258] config.c: Parsing /etc/asterisk/logger.conf [Jan 17 21:32:19] VERBOSE[3258] config.c: == Found [Jan 17 21:32:19] VERBOSE[3258] logger.c: Asterisk Queue Logger restarted [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:35774 ---> PUBLISH sip:*10@192.168.3.201 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-ac3a0e3b5a2a4f3d-1---d8754z-;rport Max-Forwards: 70 Contact: To: "GéraldTest & Nickilo" From: "GéraldTest & Nickilo";tag=1b9fee7b Call-ID: MjA5ZDdhZmFiZTc5NTljNWNmNmM0Yzk3ZDUyNzFhZGI. CSeq: 1 PUBLISH Expires: 60 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/pidf+xml User-Agent: X-Lite release 1014k stamp 47051 Event: presence Content-Length: 482 openidleIdle <-------------> [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 0 [ 37]: PUBLISH sip:*10@192.168.3.201 SIP/2.0 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-ac3a0e3b5a2a4f3d-1---d8754z-;rport [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 3 [ 37]: Contact: [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 4 [ 50]: To: "GéraldTest & Nickilo" [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 5 [ 65]: From: "GéraldTest & Nickilo";tag=1b9fee7b [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 6 [ 53]: Call-ID: MjA5ZDdhZmFiZTc5NTljNWNmNmM0Yzk3ZDUyNzFhZGI. [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 7 [ 15]: CSeq: 1 PUBLISH [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 8 [ 11]: Expires: 60 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 10 [ 34]: Content-Type: application/pidf+xml [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 11 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 12 [ 15]: Event: presence [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 13 [ 19]: Content-Length: 482 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 14 [ 0]: [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 0 [482]: openidleIdle [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: --- (14 headers 1 lines) --- [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: = Looking for Call ID: MjA5ZDdhZmFiZTc5NTljNWNmNmM0Yzk3ZDUyNzFhZGI. (Checking From) --From tag 1b9fee7b --To-tag [Jan 17 21:32:29] DEBUG[3195] acl.c: For destination '192.168.3.21', our source address is '192.168.3.201'. [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.3.201:5060 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Allocating new SIP dialog for MjA5ZDdhZmFiZTc5NTljNWNmNmM0Yzk3ZDUyNzFhZGI. - PUBLISH (No RTP) [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: **** Received PUBLISH (15) - Command in SIP PUBLISH [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: <--- Transmitting (NAT) to 192.168.3.21:35774 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-ac3a0e3b5a2a4f3d-1---d8754z-;received=192.168.3.21;rport=35774 From: "GéraldTest & Nickilo";tag=1b9fee7b To: "GéraldTest & Nickilo";tag=as7a093a27 Call-ID: MjA5ZDdhZmFiZTc5NTljNWNmNmM0Yzk3ZDUyNzFhZGI. CSeq: 1 PUBLISH Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Trying to put 'SIP/2.0 489' onto UDP socket destined for 192.168.3.21:35774 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: SIP message could not be handled, bad request: MjA5ZDdhZmFiZTc5NTljNWNmNmM0Yzk3ZDUyNzFhZGI. [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:35774 ---> INVITE sip:*11@192.168.3.201 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-c93025274883a60f-1---d8754z-;rport Max-Forwards: 70 Contact: To: "*11 (Softphone)" From: "GéraldTest & Nickilo";tag=337d2618 Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1014k stamp 47051 Content-Length: 212 v=0 o=- 8 2 IN IP4 192.168.3.21 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.3.21 t=0 0 m=audio 3606 RTP/AVP 100 8 101 a=fmtp:101 0-15 a=rtpmap:100 SPEEX/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 0 [ 36]: INVITE sip:*11@192.168.3.201 SIP/2.0 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-c93025274883a60f-1---d8754z-;rport [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 3 [ 37]: Contact: [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 4 [ 44]: To: "*11 (Softphone)" [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 5 [ 65]: From: "GéraldTest & Nickilo";tag=337d2618 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 6 [ 53]: Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 7 [ 14]: CSeq: 1 INVITE [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 8 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 10 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 11 [ 19]: Content-Length: 212 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 12 [ 0]: [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 0 [ 3]: v=0 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 1 [ 27]: o=- 8 2 IN IP4 192.168.3.21 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 2 [ 24]: s=CounterPath X-Lite 3.0 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.3.21 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 5 [ 30]: m=audio 3606 RTP/AVP 100 8 101 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 6 [ 15]: a=fmtp:101 0-15 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 7 [ 24]: a=rtpmap:100 SPEEX/16000 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: --- (12 headers 10 lines) --- [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: = Looking for Call ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. (Checking From) --From tag 337d2618 --To-tag [Jan 17 21:32:29] DEBUG[3195] acl.c: For destination '192.168.3.21', our source address is '192.168.3.201'. [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.3.201:5060 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Allocating new SIP dialog for ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. - INVITE (No RTP) [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 17 21:32:29] DEBUG[3195] netsock2.c: Splitting '192.168.3.21:35774' gives... [Jan 17 21:32:29] DEBUG[3195] netsock2.c: ...host '192.168.3.21' and port '35774'. [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Sending to 192.168.3.21:35774 (NAT) [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Initializing initreq for method INVITE - callid ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Using INVITE request as basis request - ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Found peer '*10' for '*10' from 192.168.3.21:35774 [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.3.21:35774 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-c93025274883a60f-1---d8754z-;received=192.168.3.21;rport=35774 From: "GéraldTest & Nickilo";tag=337d2618 To: "*11 (Softphone)";tag=as53e6467c Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 1 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0d81cad1" Content-Length: 0 <------------> [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #98 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.3.21:35774 [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Scheduling destruction of SIP dialog 'ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc.' in 6464 ms (Method: INVITE) [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:35774 ---> ACK sip:*11@192.168.3.201 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-c93025274883a60f-1---d8754z-;rport To: "*11 (Softphone)";tag=as53e6467c From: "GéraldTest & Nickilo";tag=337d2618 Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 1 ACK Content-Length: 0 <-------------> [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 0 [ 33]: ACK sip:*11@192.168.3.201 SIP/2.0 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-c93025274883a60f-1---d8754z-;rport [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 2 [ 59]: To: "*11 (Softphone)";tag=as53e6467c [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 3 [ 65]: From: "GéraldTest & Nickilo";tag=337d2618 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 4 [ 53]: Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: --- (7 headers 0 lines) --- [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: = Looking for Call ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. (Checking From) --From tag 337d2618 --To-tag as53e6467c [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #98 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Stopping retransmission on 'ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc.' of Response 1: Match Found [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:35774 ---> INVITE sip:*11@192.168.3.201 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-343c1338f5c8a749-1---d8754z-;rport Max-Forwards: 70 Contact: To: "*11 (Softphone)" From: "GéraldTest & Nickilo";tag=337d2618 Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1014k stamp 47051 Authorization: Digest username="*10",realm="asterisk",nonce="0d81cad1",uri="sip:*11@192.168.3.201",response="883128ab262030d16d788d7b1f063714",algorithm=MD5 Content-Length: 212 v=0 o=- 8 2 IN IP4 192.168.3.21 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.3.21 t=0 0 m=audio 3606 RTP/AVP 100 8 101 a=fmtp:101 0-15 a=rtpmap:100 SPEEX/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 0 [ 36]: INVITE sip:*11@192.168.3.201 SIP/2.0 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-343c1338f5c8a749-1---d8754z-;rport [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 3 [ 37]: Contact: [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 4 [ 44]: To: "*11 (Softphone)" [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 5 [ 65]: From: "GéraldTest & Nickilo";tag=337d2618 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 6 [ 53]: Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 7 [ 14]: CSeq: 2 INVITE [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 8 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 10 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 11 [156]: Authorization: Digest username="*10",realm="asterisk",nonce="0d81cad1",uri="sip:*11@192.168.3.201",response="883128ab262030d16d788d7b1f063714",algorithm=MD5 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 12 [ 19]: Content-Length: 212 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 13 [ 0]: [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 0 [ 3]: v=0 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 1 [ 27]: o=- 8 2 IN IP4 192.168.3.21 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 2 [ 24]: s=CounterPath X-Lite 3.0 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.3.21 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 5 [ 30]: m=audio 3606 RTP/AVP 100 8 101 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 6 [ 15]: a=fmtp:101 0-15 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 7 [ 24]: a=rtpmap:100 SPEEX/16000 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: --- (13 headers 10 lines) --- [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: = Looking for Call ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. (Checking From) --From tag 337d2618 --To-tag [Jan 17 21:32:29] DEBUG[3195] netsock2.c: Splitting '192.168.3.201' gives... [Jan 17 21:32:29] DEBUG[3195] netsock2.c: ...host '192.168.3.201' and port '(null)'. [Jan 17 21:32:29] DEBUG[3195] netsock2.c: Splitting '192.168.3.201' gives... [Jan 17 21:32:29] DEBUG[3195] netsock2.c: ...host '192.168.3.201' and port '(null)'. [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 17 21:32:29] DEBUG[3195] netsock2.c: Splitting '192.168.3.21:35774' gives... [Jan 17 21:32:29] DEBUG[3195] netsock2.c: ...host '192.168.3.21' and port '35774'. [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Sending to 192.168.3.21:35774 (no NAT) [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Initializing initreq for method INVITE - callid ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Using INVITE request as basis request - ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Found peer '*10' for '*10' from 192.168.3.21:35774 [Jan 17 21:32:29] DEBUG[3195] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xba72bc0' [Jan 17 21:32:29] DEBUG[3195] res_rtp_asterisk.c: Allocated port 13840 for RTP instance '0xba72bc0' [Jan 17 21:32:29] DEBUG[3195] rtp_engine.c: RTP instance '0xba72bc0' is setup and ready to go [Jan 17 21:32:29] DEBUG[3195] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xba762c8' [Jan 17 21:32:29] DEBUG[3195] res_rtp_asterisk.c: Allocated port 12452 for RTP instance '0xba762c8' [Jan 17 21:32:29] DEBUG[3195] rtp_engine.c: RTP instance '0xba762c8' is setup and ready to go [Jan 17 21:32:29] DEBUG[3195] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xba762c8' [Jan 17 21:32:29] DEBUG[3195] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xba72bc0' [Jan 17 21:32:29] VERBOSE[3195] netsock2.c: == Using SIP RTP CoS mark 5 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Setting NAT on RTP to On [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Setting NAT on VRTP to On [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Processing session-level SDP o=- 8 2 IN IP4 192.168.3.21... UNSUPPORTED. [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Processing session-level SDP s=CounterPath X-Lite 3.0... UNSUPPORTED. [Jan 17 21:32:29] DEBUG[3195] netsock2.c: Splitting '192.168.3.21' gives... [Jan 17 21:32:29] DEBUG[3195] netsock2.c: ...host '192.168.3.21' and port '(null)'. [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.3.21... OK. [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Found RTP audio format 100 [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Found RTP audio format 8 [Jan 17 21:32:29] DEBUG[3195] rtp_engine.c: Setting payload 8 based on m type on 0xb34e46e0 [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Found RTP audio format 101 [Jan 17 21:32:29] DEBUG[3195] rtp_engine.c: Setting payload 101 based on m type on 0xb34e46e0 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Found audio description format SPEEX for ID 100 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:100 SPEEX/16000... OK. [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 17 21:32:29] DEBUG[3195] rtp_engine.c: Incorporating payload 8 on 0xb34e46e0 [Jan 17 21:32:29] DEBUG[3195] rtp_engine.c: Incorporating payload 100 on 0xb34e46e0 [Jan 17 21:32:29] DEBUG[3195] rtp_engine.c: Incorporating payload 101 on 0xb34e46e0 [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Capabilities: us - 0x80008 (alaw|h263), peer - audio=0x200000008 (alaw|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) [Jan 17 21:32:29] DEBUG[3195] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xba72bc0' [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Peer audio RTP is at port 192.168.3.21:3606 [Jan 17 21:32:29] DEBUG[3195] rtp_engine.c: Copying payload 8 from 0xb34e46e0 to 0xba72d6c [Jan 17 21:32:29] DEBUG[3195] rtp_engine.c: Copying payload 100 from 0xb34e46e0 to 0xba72d6c [Jan 17 21:32:29] DEBUG[3195] rtp_engine.c: Copying payload 101 from 0xb34e46e0 to 0xba72d6c [Jan 17 21:32:29] DEBUG[3195] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xba762c8' [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Peer doesn't provide video [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Checking SIP call limits for device *10 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Updating call counter for incoming call [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: Looking for *11 in local (domain 192.168.3.201) [Jan 17 21:32:29] VERBOSE[3170] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 21:32:29.673826','CHAN_START','','GéraldTest & Nickilo','*10','','','','*11','local','SIP/*10-00000002','','','','','1295296349.2','1295296349.2',3,'','')] [Jan 17 21:32:29] DEBUG[3170] cel_pgsql.c: inserting a CEL record. [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: *** Our capabilities are 0x80008 (alaw|h263) [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: This channel can handle video! HOLLYWOOD next! [Jan 17 21:32:29] DEBUG[3195] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Jan 17 21:32:29] DEBUG[3195] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: build_route: Contact hop: [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: list_route: hop: [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: SIP/*10-00000002: New call is still down.... Trying... [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: <--- Transmitting (no NAT) to 192.168.3.21:35774 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-343c1338f5c8a749-1---d8754z-;received=192.168.3.21;rport=35774 From: "GéraldTest & Nickilo";tag=337d2618 To: "*11 (Softphone)" Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.3.21:35774 [Jan 17 21:32:29] DEBUG[3175] devicestate.c: No provider found, checking channel drivers for SIP - *10 [Jan 17 21:32:29] DEBUG[3175] chan_sip.c: Checking device state for peer *10 [Jan 17 21:32:29] DEBUG[3175] devicestate.c: Changing state for SIP/*10 - state 1 (Not in use) [Jan 17 21:32:29] DEBUG[3175] devicestate.c: device 'SIP/*10' state '1' [Jan 17 21:32:29] DEBUG[3210] app_queue.c: Device 'SIP/*10' changed to state '1' (Not in use) [Jan 17 21:32:29] DEBUG[3259] pbx.c: Result of 'EXTEN' is '*11' [Jan 17 21:32:29] DEBUG[3259] pbx.c: Launching 'Dial' [Jan 17 21:32:29] VERBOSE[3259] pbx.c: -- Executing [*11@local:1] Dial("SIP/*10-00000002", "SIP/*11,60,tTkK") in new stack [Jan 17 21:32:29] VERBOSE[3170] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 21:32:29.691732','APP_START','','GéraldTest & Nickilo','*10','*10','','*11','*11','local','SIP/*10-00000002','Dial','SIP/*11,60,tTkK','','','1295296349.2','1295296349.2',3,'','')] [Jan 17 21:32:29] DEBUG[3170] cel_pgsql.c: inserting a CEL record. [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Allocating new SIP dialog for 223ce651493ead3d33ec4b1a3ede8cc6@192.168.3.201:0 - INVITE (No RTP) [Jan 17 21:32:29] DEBUG[3259] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xba67ce0' [Jan 17 21:32:29] DEBUG[3259] res_rtp_asterisk.c: Allocated port 14600 for RTP instance '0xba67ce0' [Jan 17 21:32:29] DEBUG[3259] rtp_engine.c: RTP instance '0xba67ce0' is setup and ready to go [Jan 17 21:32:29] DEBUG[3259] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xba632c0' [Jan 17 21:32:29] DEBUG[3259] res_rtp_asterisk.c: Allocated port 18228 for RTP instance '0xba632c0' [Jan 17 21:32:29] DEBUG[3259] rtp_engine.c: RTP instance '0xba632c0' is setup and ready to go [Jan 17 21:32:29] DEBUG[3259] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xba632c0' [Jan 17 21:32:29] DEBUG[3259] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xba67ce0' [Jan 17 21:32:29] VERBOSE[3259] netsock2.c: == Using SIP RTP CoS mark 5 [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Setting NAT on RTP to On [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Setting NAT on VRTP to On [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jan 17 21:32:29] DEBUG[3259] acl.c: For destination '192.168.3.21', our source address is '192.168.3.201'. [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.3.201:5060 [Jan 17 21:32:29] VERBOSE[3170] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 21:32:29.699544','CHAN_START','','','','','','','s','local','SIP/*11-00000003','','','','','1295296349.3','1295296349.2',3,'','')] [Jan 17 21:32:29] DEBUG[3170] cel_pgsql.c: inserting a CEL record. [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: *** Our capabilities are 0x80008 (alaw|h263) [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: This channel can handle video! HOLLYWOOD next! [Jan 17 21:32:29] DEBUG[3259] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Jan 17 21:32:29] DEBUG[3259] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Jan 17 21:32:29] DEBUG[3259] rtp_engine.c: Seeded SDP of 'SIP/*11-00000003' with that of 'SIP/*10-00000002' [Jan 17 21:32:29] DEBUG[3259] channel.c: Not copying variable DIALEDTIME. [Jan 17 21:32:29] DEBUG[3259] channel.c: Not copying variable ANSWEREDTIME. [Jan 17 21:32:29] DEBUG[3259] channel.c: Not copying variable DIALEDPEERNAME. [Jan 17 21:32:29] DEBUG[3259] channel.c: Not copying variable DIALEDPEERNUMBER. [Jan 17 21:32:29] DEBUG[3259] channel.c: Not copying variable DIALSTATUS. [Jan 17 21:32:29] DEBUG[3259] channel.c: Not copying variable SIPCALLID. [Jan 17 21:32:29] DEBUG[3259] channel.c: Not copying variable SIPDOMAIN. [Jan 17 21:32:29] DEBUG[3259] channel.c: Not copying variable SIPURI. [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Outgoing Call for *11 [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Updating call counter for outgoing call [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: This call needs video offers! [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: ** Our capability: 0x80008 (alaw|h263) Video flag: False Text flag: False [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jan 17 21:32:29] VERBOSE[3259] chan_sip.c: Audio is at 5060 [Jan 17 21:32:29] VERBOSE[3259] chan_sip.c: Video is at 192.168.3.201:5060 [Jan 17 21:32:29] VERBOSE[3259] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Jan 17 21:32:29] VERBOSE[3259] chan_sip.c: Adding video codec 0x80000 (h263) to SDP [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: -- Done with adding codecs to SDP [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Done building SDP. Settling with this capability: 0x80008 (alaw|h263) [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Initializing initreq for method INVITE - callid 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Header 0 [ 67]: INVITE sip:*11@192.168.3.21:5060;rinstance=9ff022a91c06da91 SIP/2.0 [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK4d461d7f [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Header 3 [ 72]: From: "G%C3%A9raldTest & Nickilo" ;tag=as77bfdb5c [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Header 4 [ 58]: To: [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Header 5 [ 37]: Contact: [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Header 6 [ 60]: Call-ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Header 9 [ 35]: Date: Mon, 17 Jan 2011 20:32:29 GMT [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jan 17 21:32:29] VERBOSE[3259] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.3.21:5060: INVITE sip:*11@192.168.3.21:5060;rinstance=9ff022a91c06da91 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK4d461d7f Max-Forwards: 70 From: "G%C3%A9raldTest & Nickilo" ;tag=as77bfdb5c To: Contact: Call-ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 17 Jan 2011 20:32:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 246 v=0 o=root 1796199063 1796199063 IN IP4 192.168.3.201 s=Asterisk PBX c=IN IP4 192.168.3.201 b=CT:384 t=0 0 m=audio 14600 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv m=video 18228 RTP/AVP 34 a=rtpmap:34 H263/90000 a=sendrecv --- [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #101 [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.3.21:5060 [Jan 17 21:32:29] VERBOSE[3259] app_dial.c: -- Called *11 [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK4d461d7f To: From: "G%C3%A9raldTest & Nickilo" ;tag=as77bfdb5c Call-ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 CSeq: 102 INVITE Content-Length: 0 <-------------> [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK4d461d7f [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 2 [ 58]: To: [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 3 [ 72]: From: "G%C3%A9raldTest & Nickilo" ;tag=as77bfdb5c [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 4 [ 60]: Call-ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: --- (7 headers 0 lines) --- [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: = Looking for Call ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 (Checking To) --From tag as77bfdb5c --To-tag [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: *** SIP TIMER: Cancelling retransmission #101 - INVITE (got response) [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060' Request 102: Found [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: SIP response 100 to standard invite [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK4d461d7f Contact: To: ;tag=e2a0084d From: "G%C3%A9raldTest & Nickilo";tag=as77bfdb5c Call-ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 CSeq: 102 INVITE User-Agent: X-Lite Beta release 4.0 v3 stamp 55153 Content-Length: 0 <-------------> [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK4d461d7f [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 2 [ 63]: Contact: [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 3 [ 71]: To: ;tag=e2a0084d [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 4 [ 71]: From: "G%C3%A9raldTest & Nickilo";tag=as77bfdb5c [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 5 [ 60]: Call-ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 7 [ 50]: User-Agent: X-Lite Beta release 4.0 v3 stamp 55153 [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 17 21:32:29] VERBOSE[3195] chan_sip.c: --- (9 headers 0 lines) --- [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: = Looking for Call ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 (Checking To) --From tag as77bfdb5c --To-tag e2a0084d [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060' Request 102: Found [Jan 17 21:32:29] DEBUG[3195] chan_sip.c: SIP response 180 to standard invite [Jan 17 21:32:29] DEBUG[3175] devicestate.c: No provider found, checking channel drivers for SIP - *11 [Jan 17 21:32:29] DEBUG[3175] chan_sip.c: Checking device state for peer *11 [Jan 17 21:32:29] DEBUG[3175] devicestate.c: Changing state for SIP/*11 - state 1 (Not in use) [Jan 17 21:32:29] DEBUG[3175] devicestate.c: device 'SIP/*11' state '1' [Jan 17 21:32:29] DEBUG[3210] app_queue.c: Device 'SIP/*11' changed to state '1' (Not in use) [Jan 17 21:32:29] VERBOSE[3259] app_dial.c: -- SIP/*11-00000003 is ringing [Jan 17 21:32:29] DEBUG[3259] rtp_engine.c: Setting early bridge SDP of 'SIP/*10-00000002' with that of 'SIP/*11-00000003' [Jan 17 21:32:29] VERBOSE[3259] chan_sip.c: <--- Transmitting (no NAT) to 192.168.3.21:35774 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-343c1338f5c8a749-1---d8754z-;received=192.168.3.21;rport=35774 From: "GéraldTest & Nickilo";tag=337d2618 To: "*11 (Softphone)";tag=as7b767c31 Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jan 17 21:32:29] DEBUG[3259] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.3.21:35774 [Jan 17 21:32:33] DEBUG[3259] res_rtp_asterisk.c: RTCP NAT: Got RTCP from other end. Now sending to address 192.168.3.21:5063 [Jan 17 21:32:33] DEBUG[3259] res_rtp_asterisk.c: Got RTCP report of 132 bytes [Jan 17 21:32:33] DEBUG[3259] res_rtp_asterisk.c: RTCP NAT: Got RTCP from other end. Now sending to address 192.168.3.21:5065 [Jan 17 21:32:33] DEBUG[3259] res_rtp_asterisk.c: Got RTCP report of 132 bytes [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK4d461d7f Contact: To: ;tag=e2a0084d From: "G%C3%A9raldTest & Nickilo";tag=as77bfdb5c Call-ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite Beta release 4.0 v3 stamp 55153 Content-Length: 211 v=0 o=- 1 2 IN IP4 192.168.3.21 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.3.21 t=0 0 m=audio 5062 RTP/AVP 8 a=sendrecv m=video 5064 RTP/AVP 34 a=inactive a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=3 <-------------> [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK4d461d7f [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 2 [ 63]: Contact: [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 3 [ 71]: To: ;tag=e2a0084d [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 4 [ 71]: From: "G%C3%A9raldTest & Nickilo";tag=as77bfdb5c [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 5 [ 60]: Call-ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 9 [ 50]: User-Agent: X-Lite Beta release 4.0 v3 stamp 55153 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 10 [ 19]: Content-Length: 211 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 11 [ 0]: [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Body 0 [ 3]: v=0 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Body 1 [ 27]: o=- 1 2 IN IP4 192.168.3.21 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Body 2 [ 24]: s=CounterPath X-Lite 4.0 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.3.21 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Body 5 [ 22]: m=audio 5062 RTP/AVP 8 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Body 6 [ 10]: a=sendrecv [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Body 7 [ 23]: m=video 5064 RTP/AVP 34 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Body 8 [ 10]: a=inactive [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Body 9 [ 22]: a=rtpmap:34 H263/90000 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Body 10 [ 22]: a=fmtp:34 QCIF=2;CIF=3 [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: --- (11 headers 11 lines) --- [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: = Looking for Call ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 (Checking To) --From tag as77bfdb5c --To-tag e2a0084d [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Acked pending invite 102 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Stopping retransmission on '3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060' of Request 102: Match Found [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: SIP response 200 to standard invite [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Processing session-level SDP o=- 1 2 IN IP4 192.168.3.21... UNSUPPORTED. [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Processing session-level SDP s=CounterPath X-Lite 4.0... UNSUPPORTED. [Jan 17 21:32:33] DEBUG[3195] netsock2.c: Splitting '192.168.3.21' gives... [Jan 17 21:32:33] DEBUG[3195] netsock2.c: ...host '192.168.3.21' and port '(null)'. [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.3.21... OK. [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: Found RTP audio format 8 [Jan 17 21:32:33] DEBUG[3195] rtp_engine.c: Setting payload 8 based on m type on 0xb34e4d30 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: Found RTP video format 34 [Jan 17 21:32:33] DEBUG[3195] rtp_engine.c: Setting payload 34 based on m type on 0xb34e40b0 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Processing media-level (video) SDP a=inactive... OK. [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: Found video description format H263 for ID 34 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Processing media-level (video) SDP a=rtpmap:34 H263/90000... OK. [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Processing media-level (video) SDP a=fmtp:34 QCIF=2;CIF=3... UNSUPPORTED. [Jan 17 21:32:33] DEBUG[3195] rtp_engine.c: Incorporating payload 8 on 0xb34e4d30 [Jan 17 21:32:33] DEBUG[3195] rtp_engine.c: Incorporating payload 34 on 0xb34e40b0 [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: Capabilities: us - 0x80008 (alaw|h263), peer - audio=0x8 (alaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x80008 (alaw|h263) [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [Jan 17 21:32:33] DEBUG[3195] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xba67ce0' [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: Peer audio RTP is at port 192.168.3.21:5062 [Jan 17 21:32:33] DEBUG[3195] rtp_engine.c: Copying payload 8 from 0xb34e4d30 to 0xba67e8c [Jan 17 21:32:33] DEBUG[3195] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xba632c0' [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: Peer video RTP is at port 192.168.3.21:5064 [Jan 17 21:32:33] DEBUG[3195] rtp_engine.c: Copying payload 34 from 0xb34e40b0 to 0xba6346c [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: We're settling with these formats: 0x80008 (alaw|h263) [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: We have an owner, now see if we need to change this call [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Updating call counter for outgoing call [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: build_route: Contact hop: [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: list_route: hop: [Jan 17 21:32:33] DEBUG[3195] netsock2.c: Splitting '192.168.3.21:5060' gives... [Jan 17 21:32:33] DEBUG[3195] netsock2.c: ...host '192.168.3.21' and port '5060'. [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Strict routing enforced for session 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 17 21:32:33] DEBUG[3195] netsock2.c: Splitting '192.168.3.21:5060' gives... [Jan 17 21:32:33] DEBUG[3195] netsock2.c: ...host '192.168.3.21' and port '5060'. [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: set_destination: set destination to 192.168.3.21:5060 [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: Transmitting (no NAT) to 192.168.3.21:5060: ACK sip:*11@192.168.3.21:5060;rinstance=9ff022a91c06da91 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK5db7a2a0 Max-Forwards: 70 From: "G%C3%A9raldTest & Nickilo" ;tag=as77bfdb5c To: ;tag=e2a0084d Contact: Call-ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Trying to put 'ACK sip:*11' onto UDP socket destined for 192.168.3.21:5060 [Jan 17 21:32:33] VERBOSE[3259] app_dial.c: -- SIP/*11-00000003 answered SIP/*10-00000002 [Jan 17 21:32:33] DEBUG[3259] chan_sip.c: SIP answering channel: SIP/*10-00000002 [Jan 17 21:32:33] DEBUG[3259] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jan 17 21:32:33] DEBUG[3259] chan_sip.c: Setting framing from config on incoming call [Jan 17 21:32:33] DEBUG[3259] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jan 17 21:32:33] DEBUG[3259] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jan 17 21:32:33] VERBOSE[3259] chan_sip.c: Audio is at 5060 [Jan 17 21:32:33] VERBOSE[3259] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Jan 17 21:32:33] DEBUG[3259] chan_sip.c: -- Done with adding codecs to SDP [Jan 17 21:32:33] DEBUG[3259] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jan 17 21:32:33] VERBOSE[3259] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.3.21:35774 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-343c1338f5c8a749-1---d8754z-;received=192.168.3.21;rport=35774 From: "GéraldTest & Nickilo";tag=337d2618 To: "*11 (Softphone)";tag=as7b767c31 Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 172 v=0 o=root 750788583 750788583 IN IP4 192.168.3.201 s=Asterisk PBX c=IN IP4 192.168.3.201 t=0 0 m=audio 13840 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv <------------> [Jan 17 21:32:33] DEBUG[3259] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #104 [Jan 17 21:32:33] DEBUG[3259] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.3.21:35774 [Jan 17 21:32:33] DEBUG[3259] features.c: bridge answer set, chan answer set [Jan 17 21:32:33] DEBUG[3259] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 17 21:32:33] DEBUG[3259] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 17 21:32:33] VERBOSE[3259] rtp_engine.c: -- Locally bridging SIP/*10-00000002 and SIP/*11-00000003 [Jan 17 21:32:33] DEBUG[3175] devicestate.c: No provider found, checking channel drivers for SIP - *11 [Jan 17 21:32:33] DEBUG[3175] chan_sip.c: Checking device state for peer *11 [Jan 17 21:32:33] DEBUG[3175] devicestate.c: Changing state for SIP/*11 - state 1 (Not in use) [Jan 17 21:32:33] DEBUG[3175] devicestate.c: device 'SIP/*11' state '1' [Jan 17 21:32:33] DEBUG[3175] devicestate.c: No provider found, checking channel drivers for SIP - *10 [Jan 17 21:32:33] DEBUG[3175] chan_sip.c: Checking device state for peer *10 [Jan 17 21:32:33] DEBUG[3175] devicestate.c: Changing state for SIP/*10 - state 1 (Not in use) [Jan 17 21:32:33] DEBUG[3175] devicestate.c: device 'SIP/*10' state '1' [Jan 17 21:32:33] VERBOSE[3170] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 21:32:33.868399','ANSWER','','','*11','','','','*11','local','SIP/*11-00000003','AppDial','(Outgoing Line)','','','1295296349.3','1295296349.2',3,'','')] [Jan 17 21:32:33] DEBUG[3170] cel_pgsql.c: inserting a CEL record. [Jan 17 21:32:33] DEBUG[3259] res_rtp_asterisk.c: Got RTCP report of 132 bytes [Jan 17 21:32:33] VERBOSE[3170] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 21:32:33.869857','ANSWER','','GéraldTest & Nickilo','*10','*10','','*11','*11','local','SIP/*10-00000002','Dial','SIP/*11,60,tTkK','','','1295296349.2','1295296349.2',3,'','')] [Jan 17 21:32:33] DEBUG[3170] cel_pgsql.c: inserting a CEL record. [Jan 17 21:32:33] VERBOSE[3170] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 21:32:33.870096','BRIDGE_START','','GéraldTest & Nickilo','*10','*10','','*11','*11','local','SIP/*10-00000002','Dial','SIP/*11,60,tTkK','','','1295296349.2','1295296349.2',3,'','')] [Jan 17 21:32:33] DEBUG[3170] cel_pgsql.c: inserting a CEL record. [Jan 17 21:32:33] DEBUG[3210] app_queue.c: Device 'SIP/*11' changed to state '1' (Not in use) [Jan 17 21:32:33] DEBUG[3210] app_queue.c: Device 'SIP/*10' changed to state '1' (Not in use) [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc.' Method: INVITE [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060' Method: INVITE [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: SIP TIMER: Rescheduling retransmission #104 (1) SIP/2.0 - 1 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 202 ms (t1 101 ms (Retrans id #104)) [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: Retransmitting #1 (no NAT) to 192.168.3.21:35774: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-343c1338f5c8a749-1---d8754z-;received=192.168.3.21;rport=35774 From: "GéraldTest & Nickilo";tag=337d2618 To: "*11 (Softphone)";tag=as7b767c31 Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 172 v=0 o=root 750788583 750788583 IN IP4 192.168.3.201 s=Asterisk PBX c=IN IP4 192.168.3.201 t=0 0 m=audio 13840 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv --- [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.3.21:35774 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc.' Method: INVITE [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060' Method: INVITE [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:35774 ---> ACK sip:*11@192.168.3.201:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-5338993f2977416d-1---d8754z-;rport Max-Forwards: 70 Contact: To: "*11 (Softphone)";tag=as7b767c31 From: "GéraldTest & Nickilo";tag=337d2618 Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 2 ACK User-Agent: X-Lite release 1014k stamp 47051 Authorization: Digest username="*10",realm="asterisk",nonce="0d81cad1",uri="sip:*11@192.168.3.201",response="883128ab262030d16d788d7b1f063714",algorithm=MD5 Content-Length: 0 <-------------> [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 0 [ 38]: ACK sip:*11@192.168.3.201:5060 SIP/2.0 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-5338993f2977416d-1---d8754z-;rport [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 3 [ 37]: Contact: [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 4 [ 59]: To: "*11 (Softphone)";tag=as7b767c31 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 5 [ 65]: From: "GéraldTest & Nickilo";tag=337d2618 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 6 [ 53]: Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 7 [ 11]: CSeq: 2 ACK [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 8 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 9 [156]: Authorization: Digest username="*10",realm="asterisk",nonce="0d81cad1",uri="sip:*11@192.168.3.201",response="883128ab262030d16d788d7b1f063714",algorithm=MD5 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: --- (11 headers 0 lines) --- [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: = Looking for Call ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. (Checking From) --From tag 337d2618 --To-tag as7b767c31 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #104 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Stopping retransmission on 'ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc.' of Response 2: Match Found [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc.' Method: ACK [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060' Method: INVITE [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:35774 ---> ACK sip:*11@192.168.3.201:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-5338993f2977416d-1---d8754z-;rport Max-Forwards: 70 Contact: To: "*11 (Softphone)";tag=as7b767c31 From: "GéraldTest & Nickilo";tag=337d2618 Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 2 ACK User-Agent: X-Lite release 1014k stamp 47051 Authorization: Digest username="*10",realm="asterisk",nonce="0d81cad1",uri="sip:*11@192.168.3.201",response="883128ab262030d16d788d7b1f063714",algorithm=MD5 Content-Length: 0 <-------------> [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 0 [ 38]: ACK sip:*11@192.168.3.201:5060 SIP/2.0 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-5338993f2977416d-1---d8754z-;rport [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 3 [ 37]: Contact: [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 4 [ 59]: To: "*11 (Softphone)";tag=as7b767c31 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 5 [ 65]: From: "GéraldTest & Nickilo";tag=337d2618 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 6 [ 53]: Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 7 [ 11]: CSeq: 2 ACK [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 8 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 9 [156]: Authorization: Digest username="*10",realm="asterisk",nonce="0d81cad1",uri="sip:*11@192.168.3.201",response="883128ab262030d16d788d7b1f063714",algorithm=MD5 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jan 17 21:32:33] VERBOSE[3195] chan_sip.c: --- (11 headers 0 lines) --- [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: = Looking for Call ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. (Checking From) --From tag 337d2618 --To-tag as7b767c31 [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc.' Method: ACK [Jan 17 21:32:33] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060' Method: INVITE [Jan 17 21:32:34] NOTICE[3259] res_rtp_asterisk.c: Unknown RTP codec 126 received from '192.168.3.21:5064' [Jan 17 21:32:34] NOTICE[3259] res_rtp_asterisk.c: Unknown RTP codec 126 received from '192.168.3.21:5064' [Jan 17 21:32:34] NOTICE[3259] res_rtp_asterisk.c: Unknown RTP codec 126 received from '192.168.3.21:5064' [Jan 17 21:32:34] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc.' Method: ACK [Jan 17 21:32:34] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060' Method: INVITE [Jan 17 21:32:35] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc.' Method: ACK [Jan 17 21:32:35] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060' Method: INVITE [Jan 17 21:32:36] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:5060 ---> <-------------> [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Header 0 [ 0]: [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc.' Method: ACK [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060' Method: INVITE [Jan 17 21:32:36] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:35774 ---> INFO sip:*11@192.168.3.201:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-7f8f2d3ba6a44e5b-1---d8754z-;rport Max-Forwards: 70 Contact: To: "*11 (Softphone)";tag=as7b767c31 From: "GéraldTest & Nickilo";tag=337d2618 Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 3 INFO Content-Type: application/dtmf-relay User-Agent: X-Lite release 1014k stamp 47051 Authorization: Digest username="*10",realm="asterisk",nonce="0d81cad1",uri="sip:*11@192.168.3.201:5060",response="414d055473fb551434cbec39b4e25a39",algorithm=MD5 Content-Length: 22 Signal=# Duration=250 <-------------> [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Header 0 [ 39]: INFO sip:*11@192.168.3.201:5060 SIP/2.0 [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-7f8f2d3ba6a44e5b-1---d8754z-;rport [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Header 3 [ 37]: Contact: [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Header 4 [ 59]: To: "*11 (Softphone)";tag=as7b767c31 [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Header 5 [ 65]: From: "GéraldTest & Nickilo";tag=337d2618 [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Header 6 [ 53]: Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Header 7 [ 12]: CSeq: 3 INFO [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Header 8 [ 36]: Content-Type: application/dtmf-relay [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Header 9 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Header 10 [161]: Authorization: Digest username="*10",realm="asterisk",nonce="0d81cad1",uri="sip:*11@192.168.3.201:5060",response="414d055473fb551434cbec39b4e25a39",algorithm=MD5 [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Header 11 [ 18]: Content-Length: 22 [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Header 12 [ 0]: [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Body 0 [ 8]: Signal=# [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Body 1 [ 12]: Duration=250 [Jan 17 21:32:36] VERBOSE[3195] chan_sip.c: --- (12 headers 2 lines) --- [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: = Looking for Call ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. (Checking From) --From tag 337d2618 --To-tag as7b767c31 [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: **** Received INFO (13) - Command in SIP INFO [Jan 17 21:32:36] VERBOSE[3195] chan_sip.c: Receiving INFO! [Jan 17 21:32:36] VERBOSE[3195] chan_sip.c: * DTMF-relay event received: # [Jan 17 21:32:36] VERBOSE[3195] chan_sip.c: <--- Transmitting (no NAT) to 192.168.3.21:35774 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-7f8f2d3ba6a44e5b-1---d8754z-;received=192.168.3.21;rport=35774 From: "GéraldTest & Nickilo";tag=337d2618 To: "*11 (Softphone)";tag=as7b767c31 Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 3 INFO Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.3.21:35774 [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc.' Method: INFO [Jan 17 21:32:36] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060' Method: INVITE [Jan 17 21:32:36] DTMF[3259] channel.c: DTMF end '#' received on SIP/*10-00000002, duration 250 ms [Jan 17 21:32:36] DTMF[3259] channel.c: DTMF begin emulation of '#' with duration 250 queued on SIP/*10-00000002 [Jan 17 21:32:36] DEBUG[3259] rtp_engine.c: rtp-engine-local-bridge: Ooh, got a digit [Jan 17 21:32:36] DEBUG[3259] channel.c: Returning from native bridge, channels: SIP/*10-00000002, SIP/*11-00000003 [Jan 17 21:32:36] DEBUG[3259] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 17 21:32:36] DEBUG[3259] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 17 21:32:36] VERBOSE[3259] rtp_engine.c: -- Locally bridging SIP/*10-00000002 and SIP/*11-00000003 [Jan 17 21:32:36] DEBUG[3259] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 21:32:36] DEBUG[3259] res_rtp_asterisk.c: Got RTCP report of 152 bytes [Jan 17 21:32:36] DEBUG[3259] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 21:32:36] DTMF[3259] channel.c: DTMF end emulation of '#' queued on SIP/*10-00000002 [Jan 17 21:32:36] DEBUG[3259] rtp_engine.c: rtp-engine-local-bridge: Ooh, got a digit [Jan 17 21:32:36] DEBUG[3259] channel.c: Returning from native bridge, channels: SIP/*10-00000002, SIP/*11-00000003 [Jan 17 21:32:36] DEBUG[3259] features.c: Feature interpret: chan=SIP/*10-00000002, peer=SIP/*11-00000003, code=#, sense=1, features=34, dynamic=# [Jan 17 21:32:36] DEBUG[3259] features.c: Set feature timer to 2000 [Jan 17 21:32:36] DEBUG[3259] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 17 21:32:36] DEBUG[3259] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 17 21:32:36] VERBOSE[3259] rtp_engine.c: -- Locally bridging SIP/*10-00000002 and SIP/*11-00000003 [Jan 17 21:32:37] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:35774 ---> INFO sip:*11@192.168.3.201:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-c711f720a30d8510-1---d8754z-;rport Max-Forwards: 70 Contact: To: "*11 (Softphone)";tag=as7b767c31 From: "GéraldTest & Nickilo";tag=337d2618 Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 4 INFO Content-Type: application/dtmf-relay User-Agent: X-Lite release 1014k stamp 47051 Authorization: Digest username="*10",realm="asterisk",nonce="0d81cad1",uri="sip:*11@192.168.3.201:5060",response="414d055473fb551434cbec39b4e25a39",algorithm=MD5 Content-Length: 22 Signal=0 Duration=250 <-------------> [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Header 0 [ 39]: INFO sip:*11@192.168.3.201:5060 SIP/2.0 [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-c711f720a30d8510-1---d8754z-;rport [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Header 3 [ 37]: Contact: [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Header 4 [ 59]: To: "*11 (Softphone)";tag=as7b767c31 [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Header 5 [ 65]: From: "GéraldTest & Nickilo";tag=337d2618 [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Header 6 [ 53]: Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Header 7 [ 12]: CSeq: 4 INFO [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Header 8 [ 36]: Content-Type: application/dtmf-relay [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Header 9 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Header 10 [161]: Authorization: Digest username="*10",realm="asterisk",nonce="0d81cad1",uri="sip:*11@192.168.3.201:5060",response="414d055473fb551434cbec39b4e25a39",algorithm=MD5 [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Header 11 [ 18]: Content-Length: 22 [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Header 12 [ 0]: [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Body 0 [ 8]: Signal=0 [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Body 1 [ 12]: Duration=250 [Jan 17 21:32:37] VERBOSE[3195] chan_sip.c: --- (12 headers 2 lines) --- [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: = Looking for Call ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. (Checking From) --From tag 337d2618 --To-tag as7b767c31 [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: **** Received INFO (13) - Command in SIP INFO [Jan 17 21:32:37] VERBOSE[3195] chan_sip.c: Receiving INFO! [Jan 17 21:32:37] VERBOSE[3195] chan_sip.c: * DTMF-relay event received: 0 [Jan 17 21:32:37] VERBOSE[3195] chan_sip.c: <--- Transmitting (no NAT) to 192.168.3.21:35774 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-c711f720a30d8510-1---d8754z-;received=192.168.3.21;rport=35774 From: "GéraldTest & Nickilo";tag=337d2618 To: "*11 (Softphone)";tag=as7b767c31 Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 4 INFO Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.3.21:35774 [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc.' Method: INFO [Jan 17 21:32:37] DEBUG[3195] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060' Method: INVITE [Jan 17 21:32:37] DTMF[3259] channel.c: DTMF end '0' received on SIP/*10-00000002, duration 250 ms [Jan 17 21:32:37] DTMF[3259] channel.c: DTMF begin emulation of '0' with duration 250 queued on SIP/*10-00000002 [Jan 17 21:32:37] DEBUG[3259] rtp_engine.c: rtp-engine-local-bridge: Ooh, got a digit [Jan 17 21:32:37] DEBUG[3259] channel.c: Returning from native bridge, channels: SIP/*10-00000002, SIP/*11-00000003 [Jan 17 21:32:37] DEBUG[3259] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 17 21:32:37] DEBUG[3259] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 17 21:32:37] VERBOSE[3259] rtp_engine.c: -- Locally bridging SIP/*10-00000002 and SIP/*11-00000003 [Jan 17 21:32:37] DTMF[3259] channel.c: DTMF end emulation of '0' queued on SIP/*10-00000002 [Jan 17 21:32:37] DEBUG[3259] rtp_engine.c: rtp-engine-local-bridge: Ooh, got a digit [Jan 17 21:32:37] DEBUG[3259] channel.c: Returning from native bridge, channels: SIP/*10-00000002, SIP/*11-00000003 [Jan 17 21:32:37] DEBUG[3259] features.c: Feature interpret: chan=SIP/*10-00000002, peer=SIP/*11-00000003, code=#0, sense=1, features=34, dynamic=# [Jan 17 21:32:37] DEBUG[3259] features.c: Feature detected: fname=Attended Transfer sname=atxfer exten=#0 [Jan 17 21:32:37] DEBUG[3259] features.c: Executing Attended Transfer SIP/*10-00000002, SIP/*11-00000003 (sense=1) [Jan 17 21:32:37] DEBUG[3259] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jan 17 21:32:37] VERBOSE[3259] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/*11-00000003 [Jan 17 21:32:37] DEBUG[3259] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jan 17 21:32:37] DEBUG[3259] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Jan 17 21:32:37] DEBUG[3259] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Jan 17 21:32:37] DEBUG[3259] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xba72bc0' [Jan 17 21:32:37] DEBUG[3259] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jan 17 21:32:37] VERBOSE[3259] file.c: -- Playing 'pbx-transfer.alaw' (language 'fr') [Jan 17 21:32:37] DEBUG[3229] res_musiconhold.c: SIP/*11-00000003 Opened file 0 '/var/lib/asterisk/moh/Musique Attente.mp3' [Jan 17 21:32:37] DEBUG[3229] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Jan 17 21:32:37] DEBUG[3229] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Jan 17 21:32:38] DEBUG[3259] channel.c: Scheduling timer at (54 requested / 54 actual) timer ticks per second [Jan 17 21:32:38] DEBUG[3259] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 17 21:32:38] DEBUG[3259] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 17 21:32:38] DEBUG[3259] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 17 21:32:38] DEBUG[3259] channel.c: Set channel SIP/*10-00000002 to write format slin [Jan 17 21:32:38] DEBUG[3259] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jan 17 21:32:39] DEBUG[3229] res_rtp_asterisk.c: Got RTCP report of 200 bytes [Jan 17 21:32:39] DEBUG[3229] res_rtp_asterisk.c: Got RTCP report of 156 bytes [Jan 17 21:32:39] DEBUG[3259] res_rtp_asterisk.c: Got RTCP report of 200 bytes [Jan 17 21:32:40] DEBUG[3259] res_rtp_asterisk.c: Got RTCP report of 160 bytes [Jan 17 21:32:40] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:35774 ---> BYE sip:*11@192.168.3.201:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-8e67fe10dc07ce26-1---d8754z-;rport Max-Forwards: 70 Contact: To: "*11 (Softphone)";tag=as7b767c31 From: "GéraldTest & Nickilo";tag=337d2618 Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 5 BYE User-Agent: X-Lite release 1014k stamp 47051 Authorization: Digest username="*10",realm="asterisk",nonce="0d81cad1",uri="sip:*11@192.168.3.201:5060",response="bab1dc9bfee62c5d199fb7d412ecea8d",algorithm=MD5 Reason: SIP;description="User Hung Up" Content-Length: 0 <-------------> [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 0 [ 38]: BYE sip:*11@192.168.3.201:5060 SIP/2.0 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-8e67fe10dc07ce26-1---d8754z-;rport [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 3 [ 37]: Contact: [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 4 [ 59]: To: "*11 (Softphone)";tag=as7b767c31 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 5 [ 65]: From: "GéraldTest & Nickilo";tag=337d2618 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 6 [ 53]: Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 7 [ 11]: CSeq: 5 BYE [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 8 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 9 [161]: Authorization: Digest username="*10",realm="asterisk",nonce="0d81cad1",uri="sip:*11@192.168.3.201:5060",response="bab1dc9bfee62c5d199fb7d412ecea8d",algorithm=MD5 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 10 [ 38]: Reason: SIP;description="User Hung Up" [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jan 17 21:32:40] VERBOSE[3195] chan_sip.c: --- (12 headers 0 lines) --- [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: = Looking for Call ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. (Checking From) --From tag 337d2618 --To-tag as7b767c31 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Initializing initreq for method BYE - callid ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:40] DEBUG[3195] netsock2.c: Splitting '192.168.3.21:35774' gives... [Jan 17 21:32:40] DEBUG[3195] netsock2.c: ...host '192.168.3.21' and port '35774'. [Jan 17 21:32:40] VERBOSE[3195] chan_sip.c: Sending to 192.168.3.21:35774 (no NAT) [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Setting SIP_ALREADYGONE on dialog ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:40] DEBUG[3195] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xba72bc0' [Jan 17 21:32:40] DEBUG[3195] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xba762c8' [Jan 17 21:32:40] VERBOSE[3195] chan_sip.c: Scheduling destruction of SIP dialog 'ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc.' in 6464 ms (Method: BYE) [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Received bye, issuing owner hangup [Jan 17 21:32:40] VERBOSE[3195] chan_sip.c: <--- Transmitting (no NAT) to 192.168.3.21:35774 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-8e67fe10dc07ce26-1---d8754z-;received=192.168.3.21;rport=35774 From: "GéraldTest & Nickilo";tag=337d2618 To: "*11 (Softphone)";tag=as7b767c31 Call-ID: ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. CSeq: 5 BYE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.3.21:35774 [Jan 17 21:32:40] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:35774 ---> PUBLISH sip:*10@192.168.3.201 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-1fcd761fe763f52a-1---d8754z-;rport Max-Forwards: 70 Contact: To: "GéraldTest & Nickilo" From: "GéraldTest & Nickilo";tag=168e0f28 Call-ID: ODgzNjllMTI5MTk4YmY5NzU1NmNmMTM0MDI0ZmY1MjQ. CSeq: 1 PUBLISH Expires: 60 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/pidf+xml User-Agent: X-Lite release 1014k stamp 47051 Event: presence Content-Length: 482 openidleIdle <-------------> [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 0 [ 37]: PUBLISH sip:*10@192.168.3.201 SIP/2.0 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-1fcd761fe763f52a-1---d8754z-;rport [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 3 [ 37]: Contact: [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 4 [ 50]: To: "GéraldTest & Nickilo" [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 5 [ 65]: From: "GéraldTest & Nickilo";tag=168e0f28 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 6 [ 53]: Call-ID: ODgzNjllMTI5MTk4YmY5NzU1NmNmMTM0MDI0ZmY1MjQ. [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 7 [ 15]: CSeq: 1 PUBLISH [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 8 [ 11]: Expires: 60 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 10 [ 34]: Content-Type: application/pidf+xml [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 11 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 12 [ 15]: Event: presence [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 13 [ 19]: Content-Length: 482 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 14 [ 0]: [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Body 0 [482]: openidleIdle [Jan 17 21:32:40] VERBOSE[3195] chan_sip.c: --- (14 headers 1 lines) --- [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: = Looking for Call ID: ODgzNjllMTI5MTk4YmY5NzU1NmNmMTM0MDI0ZmY1MjQ. (Checking From) --From tag 168e0f28 --To-tag [Jan 17 21:32:40] DEBUG[3195] acl.c: For destination '192.168.3.21', our source address is '192.168.3.201'. [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.3.201:5060 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Allocating new SIP dialog for ODgzNjllMTI5MTk4YmY5NzU1NmNmMTM0MDI0ZmY1MjQ. - PUBLISH (No RTP) [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: **** Received PUBLISH (15) - Command in SIP PUBLISH [Jan 17 21:32:40] VERBOSE[3195] chan_sip.c: <--- Transmitting (NAT) to 192.168.3.21:35774 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 192.168.3.21:35774;branch=z9hG4bK-d8754z-1fcd761fe763f52a-1---d8754z-;received=192.168.3.21;rport=35774 From: "GéraldTest & Nickilo";tag=168e0f28 To: "GéraldTest & Nickilo";tag=as17537dea Call-ID: ODgzNjllMTI5MTk4YmY5NzU1NmNmMTM0MDI0ZmY1MjQ. CSeq: 1 PUBLISH Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Trying to put 'SIP/2.0 489' onto UDP socket destined for 192.168.3.21:35774 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: SIP message could not be handled, bad request: ODgzNjllMTI5MTk4YmY5NzU1NmNmMTM0MDI0ZmY1MjQ. [Jan 17 21:32:40] DEBUG[3259] channel.c: Set channel SIP/*10-00000002 to write format alaw [Jan 17 21:32:40] DEBUG[3259] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 17 21:32:40] DEBUG[3259] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jan 17 21:32:40] VERBOSE[3259] res_musiconhold.c: -- Stopped music on hold on SIP/*11-00000003 [Jan 17 21:32:40] DEBUG[3259] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 17 21:32:40] VERBOSE[3170] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 21:32:40.709011','BRIDGE_END','','GéraldTest & Nickilo','*10','*10','','*11','*11','local','SIP/*10-00000002','Dial','SIP/*11,60,tTkK','','','1295296349.2','1295296349.2',3,'','')] [Jan 17 21:32:40] DEBUG[3170] cel_pgsql.c: inserting a CEL record. [Jan 17 21:32:40] DEBUG[3259] pbx.c: Launching 'Hangup' [Jan 17 21:32:40] VERBOSE[3259] pbx.c: -- Executing [h@local:1] Hangup("SIP/*10-00000002", "") in new stack [Jan 17 21:32:40] DEBUG[3259] cdr.c: Dropping CDR ! [Jan 17 21:32:40] VERBOSE[3170] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 21:32:40.715354','HANGUP','','','*11','','','','','local','SIP/*11-00000003','AppDial','(Outgoing Line)','','','1295296349.3','1295296349.2',3,'','')] [Jan 17 21:32:40] DEBUG[3170] cel_pgsql.c: inserting a CEL record. [Jan 17 21:32:40] DEBUG[3259] channel.c: Hanging up channel 'SIP/*11-00000003' [Jan 17 21:32:40] DEBUG[3259] chan_sip.c: Hangup call SIP/*11-00000003, SIP callid 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 [Jan 17 21:32:40] DEBUG[3259] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xba67ce0' [Jan 17 21:32:40] DEBUG[3259] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xba632c0' [Jan 17 21:32:40] VERBOSE[3259] chan_sip.c: Scheduling destruction of SIP dialog '3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060' in 6400 ms (Method: INVITE) [Jan 17 21:32:40] DEBUG[3259] chan_sip.c: Strict routing enforced for session 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 [Jan 17 21:32:40] VERBOSE[3259] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 17 21:32:40] DEBUG[3259] netsock2.c: Splitting '192.168.3.21:5060' gives... [Jan 17 21:32:40] DEBUG[3259] netsock2.c: ...host '192.168.3.21' and port '5060'. [Jan 17 21:32:40] VERBOSE[3259] chan_sip.c: set_destination: set destination to 192.168.3.21:5060 [Jan 17 21:32:40] VERBOSE[3259] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.3.21:5060: BYE sip:*11@192.168.3.21:5060;rinstance=9ff022a91c06da91 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK3c124dc7 Max-Forwards: 70 From: "G%C3%A9raldTest & Nickilo" ;tag=as77bfdb5c To: ;tag=e2a0084d Call-ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 CSeq: 103 BYE User-Agent: Asterisk PBX X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jan 17 21:32:40] DEBUG[3259] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #110 [Jan 17 21:32:40] DEBUG[3259] chan_sip.c: Trying to put 'BYE sip:*11' onto UDP socket destined for 192.168.3.21:5060 [Jan 17 21:32:40] VERBOSE[3170] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 21:32:40.725640','CHAN_END','','','*11','','','','','local','SIP/*11-00000003','AppDial','(Outgoing Line)','','','1295296349.3','1295296349.2',3,'','')] [Jan 17 21:32:40] DEBUG[3170] cel_pgsql.c: inserting a CEL record. [Jan 17 21:32:40] DEBUG[3175] devicestate.c: No provider found, checking channel drivers for SIP - *11 [Jan 17 21:32:40] DEBUG[3175] chan_sip.c: Checking device state for peer *11 [Jan 17 21:32:40] DEBUG[3175] devicestate.c: Changing state for SIP/*11 - state 1 (Not in use) [Jan 17 21:32:40] DEBUG[3175] devicestate.c: device 'SIP/*11' state '1' [Jan 17 21:32:40] DEBUG[3210] app_queue.c: Device 'SIP/*11' changed to state '1' (Not in use) [Jan 17 21:32:40] DEBUG[3259] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jan 17 21:32:40] DEBUG[3259] pbx.c: Spawn extension (local,*11,1) exited non-zero on 'SIP/*10-00000002' [Jan 17 21:32:40] VERBOSE[3259] pbx.c: == Spawn extension (local, *11, 1) exited non-zero on 'SIP/*10-00000002' [Jan 17 21:32:40] DEBUG[3259] channel.c: Soft-Hanging up channel 'SIP/*10-00000002' [Jan 17 21:32:40] VERBOSE[3170] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 21:32:40.734212','HANGUP','','GéraldTest & Nickilo','*10','*10','','*11','*11','local','SIP/*10-00000002','','','','','1295296349.2','1295296349.2',3,'','')] [Jan 17 21:32:40] DEBUG[3170] cel_pgsql.c: inserting a CEL record. [Jan 17 21:32:40] DEBUG[3259] channel.c: Hanging up channel 'SIP/*10-00000002' [Jan 17 21:32:40] DEBUG[3259] chan_sip.c: Hangup call SIP/*10-00000002, SIP callid ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:40] DEBUG[3259] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xba72bc0' [Jan 17 21:32:40] DEBUG[3259] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xba762c8' [Jan 17 21:32:40] VERBOSE[3170] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 21:32:40.740962','CHAN_END','','GéraldTest & Nickilo','*10','*10','','*11','*11','local','SIP/*10-00000002','','','','','1295296349.2','1295296349.2',3,'','')] [Jan 17 21:32:40] DEBUG[3170] cel_pgsql.c: inserting a CEL record. [Jan 17 21:32:40] VERBOSE[3170] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 21:32:40.745062','LINKEDID_END','','GéraldTest & Nickilo','*10','*10','','*11','*11','local','SIP/*10-00000002','','','','','1295296349.2','1295296349.2',3,'','')] [Jan 17 21:32:40] DEBUG[3170] cel_pgsql.c: inserting a CEL record. [Jan 17 21:32:40] DEBUG[3175] devicestate.c: No provider found, checking channel drivers for SIP - *10 [Jan 17 21:32:40] DEBUG[3175] chan_sip.c: Checking device state for peer *10 [Jan 17 21:32:40] DEBUG[3175] devicestate.c: Changing state for SIP/*10 - state 1 (Not in use) [Jan 17 21:32:40] DEBUG[3175] devicestate.c: device 'SIP/*10' state '1' [Jan 17 21:32:40] DEBUG[3210] app_queue.c: Device 'SIP/*10' changed to state '1' (Not in use) [Jan 17 21:32:40] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK3c124dc7 Contact: To: ;tag=e2a0084d From: "G%C3%A9raldTest & Nickilo";tag=as77bfdb5c Call-ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 CSeq: 103 BYE User-Agent: X-Lite Beta release 4.0 v3 stamp 55153 Content-Length: 0 <-------------> [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK3c124dc7 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 2 [ 63]: Contact: [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 3 [ 71]: To: ;tag=e2a0084d [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 4 [ 71]: From: "G%C3%A9raldTest & Nickilo";tag=as77bfdb5c [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 5 [ 60]: Call-ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 6 [ 13]: CSeq: 103 BYE [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 7 [ 50]: User-Agent: X-Lite Beta release 4.0 v3 stamp 55153 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 17 21:32:40] VERBOSE[3195] chan_sip.c: --- (9 headers 0 lines) --- [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: = Looking for Call ID: 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 (Checking To) --From tag as77bfdb5c --To-tag e2a0084d [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #110 [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Stopping retransmission on '3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060' of Request 103: Match Found [Jan 17 21:32:40] DEBUG[3195] chan_sip.c: Destroying SIP dialog 3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060 [Jan 17 21:32:40] VERBOSE[3195] chan_sip.c: Really destroying SIP dialog '3afed07272ad63eb11aa02f846c2f77c@192.168.3.201:5060' Method: INVITE [Jan 17 21:32:40] DEBUG[3195] rtp_engine.c: Destroyed RTP instance '0xba67ce0' [Jan 17 21:32:40] DEBUG[3195] rtp_engine.c: Destroyed RTP instance '0xba632c0' [Jan 17 21:32:43] VERBOSE[3258] asterisk.c: -- Remote UNIX connection disconnected [Jan 17 21:32:45] VERBOSE[3171] asterisk.c: -- Remote UNIX connection [Jan 17 21:32:47] DEBUG[3195] chan_sip.c: Auto destroying SIP dialog 'ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc.' [Jan 17 21:32:47] DEBUG[3195] chan_sip.c: Destroying SIP dialog ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc. [Jan 17 21:32:47] VERBOSE[3195] chan_sip.c: Really destroying SIP dialog 'ZjY1YTY2ZDllYjA3NjMxNTU2M2U3MjgwNzkzNWE1Njc.' Method: BYE [Jan 17 21:32:47] DEBUG[3195] rtp_engine.c: Destroyed RTP instance '0xba72bc0' [Jan 17 21:32:47] DEBUG[3195] rtp_engine.c: Destroyed RTP instance '0xba762c8' [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Allocating new SIP dialog for 6dce54dd54f34287128bcc3d5c841133@192.168.3.201:0 - OPTIONS (No RTP) [Jan 17 21:32:57] DEBUG[3195] acl.c: For destination '192.168.3.21', our source address is '192.168.3.201'. [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.3.201:5060 [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Initializing initreq for method OPTIONS - callid 59c79cb05c80dbdf621c9d0e108153ad@192.168.3.201:5060 [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 0 [ 69]: OPTIONS sip:*10@192.168.3.21:35774;rinstance=77f5e1de9e350a0e SIP/2.0 [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK2abe10a4 [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 3 [ 58]: From: "CallBox" ;tag=as4eaa3554 [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 4 [ 59]: To: [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 5 [ 41]: Contact: [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 6 [ 60]: Call-ID: 59c79cb05c80dbdf621c9d0e108153ad@192.168.3.201:5060 [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 9 [ 35]: Date: Mon, 17 Jan 2011 20:32:57 GMT [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 17 21:32:57] VERBOSE[3195] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.3.21:35774: OPTIONS sip:*10@192.168.3.21:35774;rinstance=77f5e1de9e350a0e SIP/2.0 Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK2abe10a4 Max-Forwards: 70 From: "CallBox" ;tag=as4eaa3554 To: Contact: Call-ID: 59c79cb05c80dbdf621c9d0e108153ad@192.168.3.201:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 17 Jan 2011 20:32:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #111 [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.3.21:35774 [Jan 17 21:32:57] VERBOSE[3195] chan_sip.c: <--- SIP read from UDP:192.168.3.21:35774 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK2abe10a4 Contact: To: ;tag=8fd2063e From: "CallBox";tag=as4eaa3554 Call-ID: 59c79cb05c80dbdf621c9d0e108153ad@192.168.3.201:5060 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1014k stamp 47051 Content-Length: 0 <-------------> [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.3.201:5060;branch=z9hG4bK2abe10a4 [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 2 [ 33]: Contact: [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 3 [ 72]: To: ;tag=8fd2063e [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 4 [ 57]: From: "CallBox";tag=as4eaa3554 [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 5 [ 60]: Call-ID: 59c79cb05c80dbdf621c9d0e108153ad@192.168.3.201:5060 [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 7 [ 23]: Accept: application/sdp [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 10 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jan 17 21:32:57] VERBOSE[3195] chan_sip.c: --- (12 headers 0 lines) --- [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: = Looking for Call ID: 59c79cb05c80dbdf621c9d0e108153ad@192.168.3.201:5060 (Checking To) --From tag as4eaa3554 --To-tag 8fd2063e [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #111 [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Stopping retransmission on '59c79cb05c80dbdf621c9d0e108153ad@192.168.3.201:5060' of Request 102: Match Found [Jan 17 21:32:57] DEBUG[3195] chan_sip.c: Destroying SIP dialog 59c79cb05c80dbdf621c9d0e108153ad@192.168.3.201:5060 [Jan 17 21:32:57] VERBOSE[3195] chan_sip.c: Really destroying SIP dialog '59c79cb05c80dbdf621c9d0e108153ad@192.168.3.201:5060' Method: OPTIONS [Jan 17 21:34:06] NOTICE[3195] chan_sip.c: Received SIP subscribe for peer without mailbox: *11 [Jan 17 21:37:06] NOTICE[3195] chan_sip.c: Received SIP subscribe for peer without mailbox: *11