[Jan 26 10:50:27] Asterisk 11.7.0~dfsg-1ubuntu1 built by buildd @ lamiak on a x86_64 running Linux on 2013-12-24 06:02:10 UTC [Jan 26 10:50:27] DEBUG[1271] config.c: Parsing /etc/asterisk/logger.conf [Jan 26 10:50:27] VERBOSE[1271] config.c: == Parsing '/etc/asterisk/logger.conf': Found [Jan 26 10:50:27] VERBOSE[1271] logger.c: Asterisk Queue Logger restarted [Jan 26 10:50:49] DEBUG[1167] chan_sip.c: Auto destroying SIP dialog '54b68eadce52-svlcjjtzwrjq' [Jan 26 10:50:49] DEBUG[1167] chan_sip.c: Destroying SIP dialog 54b68eadce52-svlcjjtzwrjq [Jan 26 10:50:49] VERBOSE[1167] chan_sip.c: Really destroying SIP dialog '54b68eadce52-svlcjjtzwrjq' Method: REGISTER [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: ˙<--- SIP read from UDP:#PhonesPublicIP#:2048 ---> ˙REGISTER sip:#HOSTNAME# SIP/2.0 ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-ke0qzrax9ha6;rport ˙From: "799 asterisk11" ;tag=i546vd4n49 ˙To: "799 asterisk11" ˙Call-ID: 54b68eadce52-svlcjjtzwrjq ˙CSeq: 1552 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: snom300/8.7.3.25.5 ˙Allow-Events: dialog ˙X-Real-IP: 192.168.1.41 ˙Supported: path, gruu ˙Expires: 3600 ˙Content-Length: 0 ˙ ˙<-------------> [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 0 [ 52]: REGISTER sip:#HOSTNAME# SIP/2.0 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-ke0qzrax9ha6;rport [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 2 [ 79]: From: "799 asterisk11" ;tag=i546vd4n49 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 3 [ 62]: To: "799 asterisk11" [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 4 [ 34]: Call-ID: 54b68eadce52-svlcjjtzwrjq [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 5 [ 19]: CSeq: 1552 REGISTER [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 7 [304]: 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" [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 8 [ 30]: User-Agent: snom300/8.7.3.25.5 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 10 [ 23]: X-Real-IP: 192.168.1.41 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 12 [ 13]: Expires: 3600 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: --- (14 headers 0 lines) --- [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: = Looking for Call ID: 54b68eadce52-svlcjjtzwrjq (Checking From) --From tag i546vd4n49 --To-tag [Jan 26 10:50:55] DEBUG[1167] acl.c: For destination '#PhonesPublicIP#', our source address is '192.168.122.104'. [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Target address #PhonesPublicIP#:2048 is not local, substituting externaddr [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Setting SIP_TRANSPORT_UDP with address #ServersPublicIP#:5060 [Jan 26 10:50:55] DEBUG[1167] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:50:55] DEBUG[1167] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Allocating new SIP dialog for 54b68eadce52-svlcjjtzwrjq - REGISTER (No RTP) [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Initializing initreq for method REGISTER - callid 54b68eadce52-svlcjjtzwrjq [Jan 26 10:50:55] DEBUG[1167] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:50:55] DEBUG[1167] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:50:55] DEBUG[1167] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:50:55] DEBUG[1167] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: ˙<--- Transmitting (NAT) to #PhonesPublicIP#:2048 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-ke0qzrax9ha6;received=#PhonesPublicIP#;rport=2048 ˙From: "799 asterisk11" ;tag=i546vd4n49 ˙To: "799 asterisk11" ;tag=as07a3f914 ˙Call-ID: 54b68eadce52-svlcjjtzwrjq ˙CSeq: 1552 REGISTER ˙Server: RSU Telephone Server ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="#HOSTNAME#", nonce="0a39f3a6" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: Scheduling destruction of SIP dialog '54b68eadce52-svlcjjtzwrjq' in 32000 ms (Method: REGISTER) [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: ˙<--- SIP read from UDP:#PhonesPublicIP#:2048 ---> ˙REGISTER sip:#HOSTNAME# SIP/2.0 ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-7exdt1rtsq52;rport ˙From: "799 asterisk11" ;tag=i546vd4n49 ˙To: "799 asterisk11" ˙Call-ID: 54b68eadce52-svlcjjtzwrjq ˙CSeq: 1553 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: snom300/8.7.3.25.5 ˙Allow-Events: dialog ˙X-Real-IP: 192.168.1.41 ˙Supported: path, gruu ˙Authorization: Digest username="799",realm="#HOSTNAME#",nonce="0a39f3a6",uri="sip:#HOSTNAME#",response="5a746e533ee3497a51372dbf2c6c70a8",algorithm=MD5 ˙Expires: 3600 ˙Content-Length: 0 ˙ ˙<-------------> [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 0 [ 52]: REGISTER sip:#HOSTNAME# SIP/2.0 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-7exdt1rtsq52;rport [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 2 [ 79]: From: "799 asterisk11" ;tag=i546vd4n49 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 3 [ 62]: To: "799 asterisk11" [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 4 [ 34]: Call-ID: 54b68eadce52-svlcjjtzwrjq [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 5 [ 19]: CSeq: 1553 REGISTER [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 7 [304]: 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" [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 8 [ 30]: User-Agent: snom300/8.7.3.25.5 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 10 [ 23]: X-Real-IP: 192.168.1.41 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 12 [193]: Authorization: Digest username="799",realm="#HOSTNAME#",nonce="0a39f3a6",uri="sip:#HOSTNAME#",response="5a746e533ee3497a51372dbf2c6c70a8",algorithm=MD5 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: --- (15 headers 0 lines) --- [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: = Looking for Call ID: 54b68eadce52-svlcjjtzwrjq (Checking From) --From tag i546vd4n49 --To-tag [Jan 26 10:50:55] DEBUG[1167] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:50:55] DEBUG[1167] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:50:55] DEBUG[1167] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:50:55] DEBUG[1167] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Initializing initreq for method REGISTER - callid 54b68eadce52-svlcjjtzwrjq [Jan 26 10:50:55] DEBUG[1167] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:50:55] DEBUG[1167] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:50:55] DEBUG[1167] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:50:55] DEBUG[1167] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: -- Registered SIP '799' at #PhonesPublicIP#:2048 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Allocating new SIP dialog for 7998f33e6188e6f07e5ea8063fa34184@#ServersPublicIP#:5060 - OPTIONS (No RTP) [Jan 26 10:50:55] DEBUG[1167] acl.c: For destination '#PhonesPublicIP#', our source address is '192.168.122.104'. [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Target address #PhonesPublicIP#:2048 is not local, substituting externaddr [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Setting SIP_TRANSPORT_UDP with address #ServersPublicIP#:5060 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: SIP call-id changed from '7998f33e6188e6f07e5ea8063fa34184@#ServersPublicIP#:5060' to '021cdb0447b3609f7bf2c875704972b5@#ServersPublicIP#:5060' [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Initializing initreq for method OPTIONS - callid 021cdb0447b3609f7bf2c875704972b5@#ServersPublicIP#:5060 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 0 [ 55]: OPTIONS sip:799@192.168.1.41:2048;line=5jw4uomf SIP/2.0 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP #ServersPublicIP#:5060;branch=z9hG4bK17977378;rport [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as4c933c0f [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 4 [ 45]: To: [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 5 [ 43]: Contact: [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 6 [ 61]: Call-ID: 021cdb0447b3609f7bf2c875704972b5@#ServersPublicIP#:5060 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 8 [ 32]: User-Agent: RSU Telephone Server [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 9 [ 35]: Date: Mon, 26 Jan 2015 09:50:55 GMT [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: Reliably Transmitting (NAT) to #PhonesPublicIP#:2048: ˙OPTIONS sip:799@192.168.1.41:2048;line=5jw4uomf SIP/2.0 ˙Via: SIP/2.0/UDP #ServersPublicIP#:5060;branch=z9hG4bK17977378;rport ˙Max-Forwards: 70 ˙From: "asterisk" ;tag=as4c933c0f ˙To: ˙Contact: ˙Call-ID: 021cdb0447b3609f7bf2c875704972b5@#ServersPublicIP#:5060 ˙CSeq: 102 OPTIONS ˙User-Agent: RSU Telephone Server ˙Date: Mon, 26 Jan 2015 09:50:55 GMT ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Length: 0 ˙ ˙ ˙--- [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #87 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: ˙<--- Transmitting (NAT) to #PhonesPublicIP#:2048 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-7exdt1rtsq52;received=#PhonesPublicIP#;rport=2048 ˙From: "799 asterisk11" ;tag=i546vd4n49 ˙To: "799 asterisk11" ;tag=as07a3f914 ˙Call-ID: 54b68eadce52-svlcjjtzwrjq ˙CSeq: 1553 REGISTER ˙Server: RSU Telephone Server ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Expires: 3600 ˙Contact: ;expires=3600 ˙Date: Mon, 26 Jan 2015 09:50:55 GMT ˙Content-Length: 0 ˙ ˙ ˙<------------> [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: Scheduling destruction of SIP dialog '54b68eadce52-svlcjjtzwrjq' in 32000 ms (Method: REGISTER) [Jan 26 10:50:55] DEBUG[1008] devicestate.c: No provider found, checking channel drivers for SIP - 799 [Jan 26 10:50:55] DEBUG[1008] chan_sip.c: Checking device state for peer 799 [Jan 26 10:50:55] DEBUG[1008] devicestate.c: Changing state for SIP/799 - state 1 (Not in use) [Jan 26 10:50:55] DEBUG[1008] devicestate.c: device 'SIP/799' state '1' [Jan 26 10:50:55] DEBUG[1196] app_queue.c: Device 'SIP/799' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: ˙<--- SIP read from UDP:#PhonesPublicIP#:2048 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP #ServersPublicIP#:5060;branch=z9hG4bK17977378;rport=5060 ˙From: "asterisk" ;tag=as4c933c0f ˙To: ;tag=qbakiazt19 ˙Call-ID: 021cdb0447b3609f7bf2c875704972b5@#ServersPublicIP#:5060 ˙CSeq: 102 OPTIONS ˙Contact: ;reg-id=1 ˙User-Agent: snom300/8.7.3.25.5 ˙Accept-Language: en ˙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 ˙Content-Length: 0 ˙ ˙<-------------> [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP #ServersPublicIP#:5060;branch=z9hG4bK17977378;rport=5060 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as4c933c0f [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 3 [ 60]: To: ;tag=qbakiazt19 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 4 [ 61]: Call-ID: 021cdb0447b3609f7bf2c875704972b5@#ServersPublicIP#:5060 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 6 [ 59]: Contact: ;reg-id=1 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 7 [ 30]: User-Agent: snom300/8.7.3.25.5 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: --- (14 headers 0 lines) --- [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: = Looking for Call ID: 021cdb0447b3609f7bf2c875704972b5@#ServersPublicIP#:5060 (Checking To) --From tag as4c933c0f --To-tag qbakiazt19 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #87 [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Stopping retransmission on '021cdb0447b3609f7bf2c875704972b5@#ServersPublicIP#:5060' of Request 102: Match Found [Jan 26 10:50:55] DEBUG[1167] chan_sip.c: Destroying SIP dialog 021cdb0447b3609f7bf2c875704972b5@#ServersPublicIP#:5060 [Jan 26 10:50:55] VERBOSE[1167] chan_sip.c: Really destroying SIP dialog '021cdb0447b3609f7bf2c875704972b5@#ServersPublicIP#:5060' Method: OPTIONS [Jan 26 10:50:59] VERBOSE[1167] chan_sip.c: ˙<--- SIP read from UDP:#PhonesPublicIP#:2048 ---> ˙INVITE sip:200@#HOSTNAME#;user=phone SIP/2.0 ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-ehk1fx7facm1;rport ˙From: "799 asterisk11" ;tag=92mfo6km16 ˙To: ˙Call-ID: 54c60e01a739-raivdcpmtrsc ˙CSeq: 1 INVITE ˙Max-Forwards: 70 ˙Contact: ;reg-id=1 ˙X-Serialnumber: 0004133BD3D8 ˙P-Key-Flags: keys="3" ˙User-Agent: snom300/8.7.3.25.5 ˙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: 488 ˙ ˙v=0 ˙o=root 1484532559 1484532559 IN IP4 192.168.1.41 ˙s=call ˙c=IN IP4 192.168.1.41 ˙t=0 0 ˙m=audio 57244 RTP/SAVP 9 0 8 3 99 108 18 101 ˙a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:h+8yrixJmeP+FKRuX+rBVWhtpZO3a27Ab4PSTKNv ˙a=rtpmap:9 G722/8000 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:8 PCMA/8000 ˙a=rtpmap:3 GSM/8000 ˙a=rtpmap:99 G726-32/8000 ˙a=rtpmap:108 AAL2-G726-32/8000 ˙a=rtpmap:18 G729/8000 ˙a=fmtp:18 annexb=no ˙a=rtpmap:101 telephone-event/8000 ˙a=fmtp:101 0-15 ˙a=ptime:20 ˙a=sendrecv ˙<-------------> [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 0 [ 65]: INVITE sip:200@#HOSTNAME#;user=phone SIP/2.0 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-ehk1fx7facm1;rport [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 2 [ 79]: From: "799 asterisk11" ;tag=92mfo6km16 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 3 [ 56]: To: [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 4 [ 34]: Call-ID: 54c60e01a739-raivdcpmtrsc [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 7 [ 59]: Contact: ;reg-id=1 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 8 [ 28]: X-Serialnumber: 0004133BD3D8 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 9 [ 21]: P-Key-Flags: keys="3" [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 10 [ 30]: User-Agent: snom300/8.7.3.25.5 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 11 [ 23]: Accept: application/sdp [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 12 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 13 [ 42]: Allow-Events: talk, hold, refer, call-info [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 14 [ 47]: Supported: timer, 100rel, replaces, from-change [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 15 [ 35]: Session-Expires: 3600;refresher=uas [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 16 [ 10]: Min-SE: 90 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 17 [ 29]: Content-Type: application/sdp [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 18 [ 19]: Content-Length: 488 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 19 [ 0]: [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 0 [ 3]: v=0 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 1 [ 48]: o=root 1484532559 1484532559 IN IP4 192.168.1.41 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 2 [ 6]: s=call [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.41 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 5 [ 44]: m=audio 57244 RTP/SAVP 9 0 8 3 99 108 18 101 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:h+8yrixJmeP+FKRuX+rBVWhtpZO3a27Ab4PSTKNv [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 7 [ 20]: a=rtpmap:9 G722/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 10 [ 19]: a=rtpmap:3 GSM/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 11 [ 24]: a=rtpmap:99 G726-32/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 12 [ 30]: a=rtpmap:108 AAL2-G726-32/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 13 [ 21]: a=rtpmap:18 G729/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 14 [ 19]: a=fmtp:18 annexb=no [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 15 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 16 [ 15]: a=fmtp:101 0-15 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 17 [ 10]: a=ptime:20 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 18 [ 10]: a=sendrecv [Jan 26 10:50:59] VERBOSE[1167] chan_sip.c: --- (19 headers 19 lines) --- [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: = Looking for Call ID: 54c60e01a739-raivdcpmtrsc (Checking From) --From tag 92mfo6km16 --To-tag [Jan 26 10:50:59] DEBUG[1167] acl.c: For destination '#PhonesPublicIP#', our source address is '192.168.122.104'. [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Target address #PhonesPublicIP#:2048 is not local, substituting externaddr [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Setting SIP_TRANSPORT_UDP with address #ServersPublicIP#:5060 [Jan 26 10:50:59] DEBUG[1167] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:50:59] DEBUG[1167] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:50:59] VERBOSE[1167] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Allocating new SIP dialog for 54c60e01a739-raivdcpmtrsc - INVITE (No RTP) [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 26 10:50:59] DEBUG[1167][C-00000005] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change" [Jan 26 10:50:59] DEBUG[1167][C-00000005] sip/reqresp_parser.c: Found SIP option: -timer- [Jan 26 10:50:59] DEBUG[1167][C-00000005] sip/reqresp_parser.c: Matched SIP option: timer [Jan 26 10:50:59] DEBUG[1167][C-00000005] sip/reqresp_parser.c: Found SIP option: -100rel- [Jan 26 10:50:59] DEBUG[1167][C-00000005] sip/reqresp_parser.c: Matched SIP option: 100rel [Jan 26 10:50:59] DEBUG[1167][C-00000005] sip/reqresp_parser.c: Found SIP option: -replaces- [Jan 26 10:50:59] DEBUG[1167][C-00000005] sip/reqresp_parser.c: Matched SIP option: replaces [Jan 26 10:50:59] DEBUG[1167][C-00000005] sip/reqresp_parser.c: Found SIP option: -from-change- [Jan 26 10:50:59] DEBUG[1167][C-00000005] sip/reqresp_parser.c: Matched SIP option: from-change [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Initializing initreq for method INVITE - callid 54c60e01a739-raivdcpmtrsc [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Using INVITE request as basis request - 54c60e01a739-raivdcpmtrsc [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found peer '799' for '799' from #PhonesPublicIP#:2048 [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: ˙<--- Reliably Transmitting (NAT) to #PhonesPublicIP#:2048 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-ehk1fx7facm1;received=#PhonesPublicIP#;rport=2048 ˙From: "799 asterisk11" ;tag=92mfo6km16 ˙To: ;tag=as29003207 ˙Call-ID: 54c60e01a739-raivdcpmtrsc ˙CSeq: 1 INVITE ˙Server: RSU Telephone Server ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="#HOSTNAME#", nonce="24958e2c" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #91 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Scheduling destruction of SIP dialog '54c60e01a739-raivdcpmtrsc' in 6400 ms (Method: INVITE) [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: SIP TIMER: Rescheduling retransmission #91 (1) SIP/2.0 - 1 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #91)) [Jan 26 10:50:59] VERBOSE[1167] chan_sip.c: Retransmitting #1 (NAT) to #PhonesPublicIP#:2048: ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-ehk1fx7facm1;received=#PhonesPublicIP#;rport=2048 ˙From: "799 asterisk11" ;tag=92mfo6km16 ˙To: ;tag=as29003207 ˙Call-ID: 54c60e01a739-raivdcpmtrsc ˙CSeq: 1 INVITE ˙Server: RSU Telephone Server ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="#HOSTNAME#", nonce="24958e2c" ˙Content-Length: 0 ˙ ˙ ˙--- [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:50:59] VERBOSE[1167] chan_sip.c: ˙<--- SIP read from UDP:#PhonesPublicIP#:2048 ---> ˙ACK sip:200@#HOSTNAME#;user=phone SIP/2.0 ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-ehk1fx7facm1;rport ˙From: "799 asterisk11" ;tag=92mfo6km16 ˙To: ;tag=as29003207 ˙Call-ID: 54c60e01a739-raivdcpmtrsc ˙CSeq: 1 ACK ˙Max-Forwards: 70 ˙Contact: ;reg-id=1 ˙Content-Length: 0 ˙ ˙<-------------> [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 0 [ 62]: ACK sip:200@#HOSTNAME#;user=phone SIP/2.0 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-ehk1fx7facm1;rport [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 2 [ 79]: From: "799 asterisk11" ;tag=92mfo6km16 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 3 [ 71]: To: ;tag=as29003207 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 4 [ 34]: Call-ID: 54c60e01a739-raivdcpmtrsc [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 7 [ 59]: Contact: ;reg-id=1 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 26 10:50:59] VERBOSE[1167] chan_sip.c: --- (9 headers 0 lines) --- [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: = Looking for Call ID: 54c60e01a739-raivdcpmtrsc (Checking From) --From tag 92mfo6km16 --To-tag as29003207 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #91 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Stopping retransmission on '54c60e01a739-raivdcpmtrsc' of Response 1: Match Found [Jan 26 10:50:59] VERBOSE[1167] chan_sip.c: ˙<--- SIP read from UDP:#PhonesPublicIP#:2048 ---> ˙INVITE sip:200@#HOSTNAME#;user=phone SIP/2.0 ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-s5i4wfrz8597;rport ˙From: "799 asterisk11" ;tag=92mfo6km16 ˙To: ˙Call-ID: 54c60e01a739-raivdcpmtrsc ˙CSeq: 2 INVITE ˙Max-Forwards: 70 ˙Contact: ;reg-id=1 ˙X-Serialnumber: 0004133BD3D8 ˙P-Key-Flags: keys="3" ˙User-Agent: snom300/8.7.3.25.5 ˙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="799",realm="#HOSTNAME#",nonce="24958e2c",uri="sip:200@#HOSTNAME#;user=phone",response="3328c7d6e974325ba87154fede80b416",algorithm=MD5 ˙Content-Type: application/sdp ˙Content-Length: 488 ˙ ˙v=0 ˙o=root 1484532559 1484532559 IN IP4 192.168.1.41 ˙s=call ˙c=IN IP4 192.168.1.41 ˙t=0 0 ˙m=audio 57244 RTP/SAVP 9 0 8 3 99 108 18 101 ˙a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:h+8yrixJmeP+FKRuX+rBVWhtpZO3a27Ab4PSTKNv ˙a=rtpmap:9 G722/8000 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:8 PCMA/8000 ˙a=rtpmap:3 GSM/8000 ˙a=rtpmap:99 G726-32/8000 ˙a=rtpmap:108 AAL2-G726-32/8000 ˙a=rtpmap:18 G729/8000 ˙a=fmtp:18 annexb=no ˙a=rtpmap:101 telephone-event/8000 ˙a=fmtp:101 0-15 ˙a=ptime:20 ˙a=sendrecv ˙<-------------> [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 0 [ 65]: INVITE sip:200@#HOSTNAME#;user=phone SIP/2.0 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-s5i4wfrz8597;rport [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 2 [ 79]: From: "799 asterisk11" ;tag=92mfo6km16 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 3 [ 56]: To: [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 4 [ 34]: Call-ID: 54c60e01a739-raivdcpmtrsc [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 5 [ 14]: CSeq: 2 INVITE [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 7 [ 59]: Contact: ;reg-id=1 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 8 [ 28]: X-Serialnumber: 0004133BD3D8 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 9 [ 21]: P-Key-Flags: keys="3" [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 10 [ 30]: User-Agent: snom300/8.7.3.25.5 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 11 [ 23]: Accept: application/sdp [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 12 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 13 [ 42]: Allow-Events: talk, hold, refer, call-info [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 14 [ 47]: Supported: timer, 100rel, replaces, from-change [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 15 [ 35]: Session-Expires: 3600;refresher=uas [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 16 [ 10]: Min-SE: 90 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 17 [208]: Authorization: Digest username="799",realm="#HOSTNAME#",nonce="24958e2c",uri="sip:200@#HOSTNAME#;user=phone",response="3328c7d6e974325ba87154fede80b416",algorithm=MD5 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 18 [ 29]: Content-Type: application/sdp [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 19 [ 19]: Content-Length: 488 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 20 [ 0]: [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 0 [ 3]: v=0 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 1 [ 48]: o=root 1484532559 1484532559 IN IP4 192.168.1.41 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 2 [ 6]: s=call [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.41 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 5 [ 44]: m=audio 57244 RTP/SAVP 9 0 8 3 99 108 18 101 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:h+8yrixJmeP+FKRuX+rBVWhtpZO3a27Ab4PSTKNv [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 7 [ 20]: a=rtpmap:9 G722/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 10 [ 19]: a=rtpmap:3 GSM/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 11 [ 24]: a=rtpmap:99 G726-32/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 12 [ 30]: a=rtpmap:108 AAL2-G726-32/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 13 [ 21]: a=rtpmap:18 G729/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 14 [ 19]: a=fmtp:18 annexb=no [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 15 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 16 [ 15]: a=fmtp:101 0-15 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 17 [ 10]: a=ptime:20 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Body 18 [ 10]: a=sendrecv [Jan 26 10:50:59] VERBOSE[1167] chan_sip.c: --- (20 headers 19 lines) --- [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: = Looking for Call ID: 54c60e01a739-raivdcpmtrsc (Checking From) --From tag 92mfo6km16 --To-tag [Jan 26 10:50:59] DEBUG[1167] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:50:59] DEBUG[1167] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:50:59] DEBUG[1167] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:50:59] DEBUG[1167] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Initializing initreq for method INVITE - callid 54c60e01a739-raivdcpmtrsc [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Using INVITE request as basis request - 54c60e01a739-raivdcpmtrsc [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found peer '799' for '799' from #PhonesPublicIP#:2048 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7fb25c019bf8' [Jan 26 10:50:59] DEBUG[1167][C-00000005] res_rtp_asterisk.c: Allocated port 19308 for RTP instance '0x7fb25c019bf8' [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: Splitting '192.168.122.104' into... [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: ...host '192.168.122.104' and port ''. [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: RTP instance '0x7fb25c019bf8' is setup and ready to go [Jan 26 10:50:59] DEBUG[1167][C-00000005] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7fb25c019bf8' [Jan 26 10:50:59] VERBOSE[1167][C-00000005] netsock2.c: == Using SIP RTP CoS mark 5 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Setting NAT on RTP to On [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing session-level SDP o=root 1484532559 1484532559 IN IP4 192.168.1.41... OK. [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED OR FAILED. [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: Splitting '192.168.1.41' into... [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: ...host '192.168.1.41' and port ''. [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.41... OK. [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found RTP audio format 9 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Setting payload 9 based on m type on 0x7fb2c4161c90 [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found RTP audio format 0 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Setting payload 0 based on m type on 0x7fb2c4161c90 [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found RTP audio format 8 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Setting payload 8 based on m type on 0x7fb2c4161c90 [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found RTP audio format 3 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Setting payload 3 based on m type on 0x7fb2c4161c90 [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found RTP audio format 99 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Setting payload 99 based on m type on 0x7fb2c4161c90 [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found RTP audio format 108 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Setting payload 108 based on m type on 0x7fb2c4161c90 [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found RTP audio format 18 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Setting payload 18 based on m type on 0x7fb2c4161c90 [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found RTP audio format 101 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Setting payload 101 based on m type on 0x7fb2c4161c90 [Jan 26 10:50:59] DEBUG[1167][C-00000005] sip/sdp_crypto.c: local_key64 3vG9FDvsAnBlfReqyrrIRckUrcmHbMeJv3JeezW0 len 40 [Jan 26 10:50:59] DEBUG[1167][C-00000005] res_srtp.c: Adding new policy for SSRC 1815486099 [Jan 26 10:50:59] DEBUG[1167][C-00000005] sip/sdp_crypto.c: SRTP policy activated [Jan 26 10:50:59] DEBUG[1167][C-00000005] sip/sdp_crypto.c: Accepting crypto tag 1 [Jan 26 10:50:59] DEBUG[1167][C-00000005] sip/sdp_crypto.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:3vG9FDvsAnBlfReqyrrIRckUrcmHbMeJv3JeezW0 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:h+8yrixJmeP+FKRuX+rBVWhtpZO3a27Ab4PSTKNv... OK. [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found audio description format G722 for ID 9 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found audio description format GSM for ID 3 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK. [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found audio description format G726-32 for ID 99 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:99 G726-32/8000... OK. [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found audio description format AAL2-G726-32 for ID 108 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:108 AAL2-G726-32/8000... OK. [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found audio description format G729 for ID 18 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... OK. [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Capabilities: us - (ulaw|alaw|g722), peer - audio=(gsm|ulaw|alaw|g726|g729|g726aal2|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g722) [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 26 10:50:59] DEBUG[1167][C-00000005] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fb25c019bf8' [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Peer audio RTP is at port 192.168.1.41:57244 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Copying payload 0 from 0x7fb2c4161c90 to 0x7fb25c019dc0 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Copying payload 3 from 0x7fb2c4161c90 to 0x7fb25c019dc0 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Copying payload 8 from 0x7fb2c4161c90 to 0x7fb25c019dc0 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Copying payload 9 from 0x7fb2c4161c90 to 0x7fb25c019dc0 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Copying payload 18 from 0x7fb2c4161c90 to 0x7fb25c019dc0 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Copying payload 99 from 0x7fb2c4161c90 to 0x7fb25c019dc0 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Copying payload 101 from 0x7fb2c4161c90 to 0x7fb25c019dc0 [Jan 26 10:50:59] DEBUG[1167][C-00000005] rtp_engine.c: Copying payload 108 from 0x7fb2c4161c90 to 0x7fb25c019dc0 [Jan 26 10:50:59] DEBUG[1167][C-00000005] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7fb25c019bf8' [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: We're settling with these formats: (ulaw|alaw|g722) [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Checking SIP call limits for device 799 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Updating call counter for incoming call [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:50:59] DEBUG[1167][C-00000005] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: Looking for 200 in localsets-common (domain #HOSTNAME#) [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: *** Our native formats are (g722) [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: *** Joint capabilities are (ulaw|alaw|g722) [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: *** Our capabilities are (ulaw|alaw|g722) [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: *** AST_CODEC_CHOOSE formats are g722 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: This channel will not be able to handle video. [Jan 26 10:50:59] DEBUG[1167][C-00000005] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Jan 26 10:50:59] DEBUG[1167][C-00000005] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: list_route: hop: [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Incoming INVITE with 'timer' option supported [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: INVITE also has "Session-Expires" header. [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Session-Expires: 3600 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Refresher: UAS [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: INVITE also has "Min-SE" header. [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Received Min-SE: 90 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Session timer started: 93 - 54c60e01a739-raivdcpmtrsc 900000ms [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: SIP/799-00000002: New call is still down.... Trying... [Jan 26 10:50:59] VERBOSE[1167][C-00000005] chan_sip.c: ˙<--- Transmitting (NAT) to #PhonesPublicIP#:2048 ---> ˙SIP/2.0 100 Trying ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-s5i4wfrz8597;received=#PhonesPublicIP#;rport=2048 ˙From: "799 asterisk11" ;tag=92mfo6km16 ˙To: ˙Call-ID: 54c60e01a739-raivdcpmtrsc ˙CSeq: 2 INVITE ˙Server: RSU Telephone Server ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Session-Expires: 1800;refresher=uas ˙Contact: ˙Content-Length: 0 ˙ ˙ ˙<------------> [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:50:59] DEBUG[1008] devicestate.c: No provider found, checking channel drivers for SIP - 799 [Jan 26 10:50:59] DEBUG[1008] chan_sip.c: Checking device state for peer 799 [Jan 26 10:50:59] DEBUG[1008] devicestate.c: Changing state for SIP/799 - state 1 (Not in use) [Jan 26 10:50:59] DEBUG[1008] devicestate.c: device 'SIP/799' state '1' [Jan 26 10:50:59] DEBUG[1490][C-00000005] pbx.c: Launching 'NoOp' [Jan 26 10:50:59] VERBOSE[1490][C-00000005] pbx.c: -- Executing [200@localsets-common:1] NoOp("SIP/799-00000002", "") in new stack [Jan 26 10:50:59] DEBUG[1490][C-00000005] pbx.c: Launching 'Playback' [Jan 26 10:50:59] VERBOSE[1490][C-00000005] pbx.c: -- Executing [200@localsets-common:2] Playback("SIP/799-00000002", "hello-world") in new stack [Jan 26 10:50:59] DEBUG[1490][C-00000005] chan_sip.c: SIP answering channel: SIP/799-00000002 [Jan 26 10:50:59] DEBUG[1490][C-00000005] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jan 26 10:50:59] DEBUG[1196] app_queue.c: Device 'SIP/799' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 26 10:50:59] DEBUG[1490][C-00000005] chan_sip.c: Setting framing from config on incoming call [Jan 26 10:50:59] DEBUG[1008] devicestate.c: No provider found, checking channel drivers for SIP - 799 [Jan 26 10:50:59] DEBUG[1008] chan_sip.c: Checking device state for peer 799 [Jan 26 10:50:59] DEBUG[1490][C-00000005] chan_sip.c: ** Our capability: (ulaw|alaw|g722) Video flag: True Text flag: True [Jan 26 10:50:59] DEBUG[1008] devicestate.c: Changing state for SIP/799 - state 1 (Not in use) [Jan 26 10:50:59] DEBUG[1008] devicestate.c: device 'SIP/799' state '1' [Jan 26 10:50:59] DEBUG[1490][C-00000005] chan_sip.c: ** Our prefcodec: (nothing) [Jan 26 10:50:59] VERBOSE[1490][C-00000005] chan_sip.c: Audio is at 19308 [Jan 26 10:50:59] DEBUG[1490][C-00000005] sip/sdp_crypto.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:3vG9FDvsAnBlfReqyrrIRckUrcmHbMeJv3JeezW0 [Jan 26 10:50:59] VERBOSE[1490][C-00000005] chan_sip.c: Adding codec 100012 (g722) to SDP [Jan 26 10:50:59] VERBOSE[1490][C-00000005] chan_sip.c: Adding codec 100004 (alaw) to SDP [Jan 26 10:50:59] VERBOSE[1490][C-00000005] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Jan 26 10:50:59] VERBOSE[1490][C-00000005] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 26 10:50:59] DEBUG[1490][C-00000005] chan_sip.c: -- Done with adding codecs to SDP [Jan 26 10:50:59] DEBUG[1490][C-00000005] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw|g722) [Jan 26 10:50:59] VERBOSE[1490][C-00000005] chan_sip.c: ˙<--- Reliably Transmitting (NAT) to #PhonesPublicIP#:2048 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-s5i4wfrz8597;received=#PhonesPublicIP#;rport=2048 ˙From: "799 asterisk11" ;tag=92mfo6km16 ˙To: ;tag=as7d222e17 ˙Call-ID: 54c60e01a739-raivdcpmtrsc ˙CSeq: 2 INVITE ˙Server: RSU Telephone Server ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Session-Expires: 1800;refresher=uas ˙Contact: ˙Content-Type: application/sdp ˙Require: timer ˙Content-Length: 381 ˙ ˙v=0 ˙o=peter-marrat 1568657363 1568657363 IN IP4 #ServersPublicIP# ˙s=RSU Telephone Server ˙c=IN IP4 #ServersPublicIP# ˙t=0 0 ˙m=audio 19308 RTP/SAVP 9 8 0 101 ˙a=rtpmap:9 G722/8000 ˙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:3vG9FDvsAnBlfReqyrrIRckUrcmHbMeJv3JeezW0 ˙ ˙<------------> [Jan 26 10:50:59] DEBUG[1490][C-00000005] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #95 [Jan 26 10:50:59] DEBUG[1490][C-00000005] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:50:59] DEBUG[1196] app_queue.c: Device 'SIP/799' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 26 10:50:59] VERBOSE[1167] chan_sip.c: ˙<--- SIP read from UDP:#PhonesPublicIP#:2048 ---> ˙ACK sip:200@#HOSTNAME#;user=phone SIP/2.0 ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-ehk1fx7facm1;rport ˙From: "799 asterisk11" ;tag=92mfo6km16 ˙To: ;tag=as29003207 ˙Call-ID: 54c60e01a739-raivdcpmtrsc ˙CSeq: 1 ACK ˙Max-Forwards: 70 ˙Contact: ;reg-id=1 ˙Content-Length: 0 ˙ ˙<-------------> [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 0 [ 62]: ACK sip:200@#HOSTNAME#;user=phone SIP/2.0 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-ehk1fx7facm1;rport [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 2 [ 79]: From: "799 asterisk11" ;tag=92mfo6km16 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 3 [ 71]: To: ;tag=as29003207 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 4 [ 34]: Call-ID: 54c60e01a739-raivdcpmtrsc [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 7 [ 59]: Contact: ;reg-id=1 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 26 10:50:59] VERBOSE[1167] chan_sip.c: --- (9 headers 0 lines) --- [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: = Looking for Call ID: 54c60e01a739-raivdcpmtrsc (Checking From) --From tag 92mfo6km16 --To-tag as29003207 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Invalid SIP message - rejected , no callid, len 426 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: SIP TIMER: Rescheduling retransmission #95 (1) SIP/2.0 - 1 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #95)) [Jan 26 10:50:59] VERBOSE[1167] chan_sip.c: Retransmitting #1 (NAT) to #PhonesPublicIP#:2048: ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-s5i4wfrz8597;received=#PhonesPublicIP#;rport=2048 ˙From: "799 asterisk11" ;tag=92mfo6km16 ˙To: ;tag=as7d222e17 ˙Call-ID: 54c60e01a739-raivdcpmtrsc ˙CSeq: 2 INVITE ˙Server: RSU Telephone Server ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Session-Expires: 1800;refresher=uas ˙Contact: ˙Content-Type: application/sdp ˙Require: timer ˙Content-Length: 381 ˙ ˙v=0 ˙o=peter-marrat 1568657363 1568657363 IN IP4 #ServersPublicIP# ˙s=RSU Telephone Server ˙c=IN IP4 #ServersPublicIP# ˙t=0 0 ˙m=audio 19308 RTP/SAVP 9 8 0 101 ˙a=rtpmap:9 G722/8000 ˙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:3vG9FDvsAnBlfReqyrrIRckUrcmHbMeJv3JeezW0 ˙ ˙--- [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:50:59] VERBOSE[1167] chan_sip.c: ˙<--- SIP read from UDP:#PhonesPublicIP#:2048 ---> ˙ACK sip:200@#ServersPublicIP#:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-ejv4ld22h42e;rport ˙From: "799 asterisk11" ;tag=92mfo6km16 ˙To: ;tag=as7d222e17 ˙Call-ID: 54c60e01a739-raivdcpmtrsc ˙CSeq: 2 ACK ˙Max-Forwards: 70 ˙Contact: ;reg-id=1 ˙Content-Length: 0 ˙ ˙<-------------> [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 0 [ 39]: ACK sip:200@#ServersPublicIP#:5060 SIP/2.0 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-ejv4ld22h42e;rport [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 2 [ 79]: From: "799 asterisk11" ;tag=92mfo6km16 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 3 [ 71]: To: ;tag=as7d222e17 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 4 [ 34]: Call-ID: 54c60e01a739-raivdcpmtrsc [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 5 [ 11]: CSeq: 2 ACK [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 7 [ 59]: Contact: ;reg-id=1 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 26 10:50:59] VERBOSE[1167] chan_sip.c: --- (9 headers 0 lines) --- [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: = Looking for Call ID: 54c60e01a739-raivdcpmtrsc (Checking From) --From tag 92mfo6km16 --To-tag as7d222e17 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #95 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: Stopping retransmission on '54c60e01a739-raivdcpmtrsc' of Response 2: Match Found [Jan 26 10:50:59] DEBUG[1490][C-00000005] res_rtp_asterisk.c: Got RTCP report of 48 bytes [Jan 26 10:50:59] VERBOSE[1167] chan_sip.c: ˙<--- SIP read from UDP:#PhonesPublicIP#:2048 ---> ˙ACK sip:200@#ServersPublicIP#:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-ejv4ld22h42e;rport ˙From: "799 asterisk11" ;tag=92mfo6km16 ˙To: ;tag=as7d222e17 ˙Call-ID: 54c60e01a739-raivdcpmtrsc ˙CSeq: 2 ACK ˙Max-Forwards: 70 ˙Contact: ;reg-id=1 ˙Content-Length: 0 ˙ ˙<-------------> [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 0 [ 39]: ACK sip:200@#ServersPublicIP#:5060 SIP/2.0 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-ejv4ld22h42e;rport [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 2 [ 79]: From: "799 asterisk11" ;tag=92mfo6km16 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 3 [ 71]: To: ;tag=as7d222e17 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 4 [ 34]: Call-ID: 54c60e01a739-raivdcpmtrsc [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 5 [ 11]: CSeq: 2 ACK [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 7 [ 59]: Contact: ;reg-id=1 [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 26 10:50:59] VERBOSE[1167] chan_sip.c: --- (9 headers 0 lines) --- [Jan 26 10:50:59] DEBUG[1167] chan_sip.c: = Looking for Call ID: 54c60e01a739-raivdcpmtrsc (Checking From) --From tag 92mfo6km16 --To-tag as7d222e17 [Jan 26 10:50:59] DEBUG[1167][C-00000005] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 26 10:50:59] DEBUG[1490][C-00000005] res_rtp_asterisk.c: 0x7fb25c01e140 -- Probation learning mode pass with source address #PhonesPublicIP#:57244 [Jan 26 10:50:59] DEBUG[1490][C-00000005] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fb25c019bf8' [Jan 26 10:50:59] DEBUG[1490][C-00000005] channel.c: Set channel SIP/799-00000002 to write format gsm [Jan 26 10:50:59] DEBUG[1490][C-00000005] res_rtp_asterisk.c: Ooh, format changed from unknown to g722 [Jan 26 10:50:59] DEBUG[1490][C-00000005] res_rtp_asterisk.c: Created smoother: format: g722 ms: 20 len: 160 [Jan 26 10:50:59] DEBUG[1490][C-00000005] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jan 26 10:50:59] VERBOSE[1490][C-00000005] file.c: -- Playing 'hello-world.gsm' (language 'en') [Jan 26 10:50:59] DEBUG[1490][C-00000005] res_rtp_asterisk.c: 0x7fb25c01e140 -- Probation learning mode pass with source address #PhonesPublicIP#:57244 [Jan 26 10:51:01] DEBUG[1490][C-00000005] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 10:51:01] DEBUG[1490][C-00000005] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 10:51:01] DEBUG[1490][C-00000005] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 10:51:01] DEBUG[1490][C-00000005] channel.c: Set channel SIP/799-00000002 to write format g722 [Jan 26 10:51:01] DEBUG[1490][C-00000005] pbx.c: Result of 'EXTEN' is '200' [Jan 26 10:51:01] DEBUG[1490][C-00000005] devicestate.c: No provider found, checking channel drivers for SIP - 200 [Jan 26 10:51:01] DEBUG[1490][C-00000005] chan_sip.c: Checking device state for peer 200 [Jan 26 10:51:01] DEBUG[1490][C-00000005] pbx.c: Function result is 'INVALID' [Jan 26 10:51:01] DEBUG[1490][C-00000005] pbx.c: Launching 'Set' [Jan 26 10:51:01] VERBOSE[1490][C-00000005] pbx.c: -- Executing [200@localsets-common:3] Set("SIP/799-00000002", "deviceState=INVALID") in new stack [Jan 26 10:51:01] DEBUG[1490][C-00000005] pbx.c: Result of 'deviceState' is 'INVALID' [Jan 26 10:51:01] DEBUG[1490][C-00000005] pbx.c: Expression result is '1' [Jan 26 10:51:01] DEBUG[1490][C-00000005] pbx.c: Launching 'GotoIf' [Jan 26 10:51:01] VERBOSE[1490][C-00000005] pbx.c: -- Executing [200@localsets-common:4] GotoIf("SIP/799-00000002", "1?numberNotDefined") in new stack [Jan 26 10:51:01] VERBOSE[1490][C-00000005] pbx.c: -- Goto (localsets-common,200,14) [Jan 26 10:51:01] DEBUG[1490][C-00000005] pbx.c: Result of 'EXTEN' is '200' [Jan 26 10:51:01] DEBUG[1490][C-00000005] pbx.c: Launching 'Verbose' [Jan 26 10:51:01] VERBOSE[1490][C-00000005] pbx.c: -- Executing [200@localsets-common:14] Verbose("SIP/799-00000002", "Number 200 is invalid: drop the call") in new stack [Jan 26 10:51:01] VERBOSE[1490][C-00000005] app_verbose.c: Number 200 is invalid: drop the call [Jan 26 10:51:01] DEBUG[1490][C-00000005] pbx.c: Launching 'Answer' [Jan 26 10:51:01] VERBOSE[1490][C-00000005] pbx.c: -- Executing [200@localsets-common:15] Answer("SIP/799-00000002", "") in new stack [Jan 26 10:51:01] DEBUG[1490][C-00000005] pbx.c: Launching 'Playback' [Jan 26 10:51:01] VERBOSE[1490][C-00000005] pbx.c: -- Executing [200@localsets-common:16] Playback("SIP/799-00000002", "custom/system-number-is-not-in-service") in new stack [Jan 26 10:51:01] DEBUG[1490][C-00000005] channel.c: Set channel SIP/799-00000002 to write format slin [Jan 26 10:51:01] DEBUG[1490][C-00000005] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jan 26 10:51:01] VERBOSE[1490][C-00000005] file.c: -- Playing 'custom/system-number-is-not-in-service.slin' (language 'en') [Jan 26 10:51:02] VERBOSE[1167] chan_sip.c: ˙<--- SIP read from UDP:#PhonesPublicIP#:2048 ---> ˙BYE sip:200@#ServersPublicIP#:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-vh87o96ge8ku;rport ˙From: "799 asterisk11" ;tag=92mfo6km16 ˙To: ;tag=as7d222e17 ˙Call-ID: 54c60e01a739-raivdcpmtrsc ˙CSeq: 3 BYE ˙Max-Forwards: 70 ˙Contact: ;reg-id=1 ˙User-Agent: snom300/8.7.3.25.5 ˙RTP-RxStat: Total_Rx_Pkts=132,Rx_Pkts=132,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 ˙RTP-TxStat: Total_Tx_Pkts=135,Tx_Pkts=135,Remote_Tx_Pkts=0 ˙Content-Length: 0 ˙ ˙<-------------> [Jan 26 10:51:02] DEBUG[1167] chan_sip.c: Header 0 [ 39]: BYE sip:200@#ServersPublicIP#:5060 SIP/2.0 [Jan 26 10:51:02] DEBUG[1167] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-vh87o96ge8ku;rport [Jan 26 10:51:02] DEBUG[1167] chan_sip.c: Header 2 [ 79]: From: "799 asterisk11" ;tag=92mfo6km16 [Jan 26 10:51:02] DEBUG[1167] chan_sip.c: Header 3 [ 71]: To: ;tag=as7d222e17 [Jan 26 10:51:02] DEBUG[1167] chan_sip.c: Header 4 [ 34]: Call-ID: 54c60e01a739-raivdcpmtrsc [Jan 26 10:51:02] DEBUG[1167] chan_sip.c: Header 5 [ 11]: CSeq: 3 BYE [Jan 26 10:51:02] DEBUG[1167] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:51:02] DEBUG[1167] chan_sip.c: Header 7 [ 59]: Contact: ;reg-id=1 [Jan 26 10:51:02] DEBUG[1167] chan_sip.c: Header 8 [ 30]: User-Agent: snom300/8.7.3.25.5 [Jan 26 10:51:02] DEBUG[1167] chan_sip.c: Header 9 [ 78]: RTP-RxStat: Total_Rx_Pkts=132,Rx_Pkts=132,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 [Jan 26 10:51:02] DEBUG[1167] chan_sip.c: Header 10 [ 58]: RTP-TxStat: Total_Tx_Pkts=135,Tx_Pkts=135,Remote_Tx_Pkts=0 [Jan 26 10:51:02] DEBUG[1167] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jan 26 10:51:02] VERBOSE[1167] chan_sip.c: --- (12 headers 0 lines) --- [Jan 26 10:51:02] DEBUG[1167] chan_sip.c: = Looking for Call ID: 54c60e01a739-raivdcpmtrsc (Checking From) --From tag 92mfo6km16 --To-tag as7d222e17 [Jan 26 10:51:02] DEBUG[1167][C-00000005] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jan 26 10:51:02] DEBUG[1167][C-00000005] chan_sip.c: Initializing initreq for method BYE - callid 54c60e01a739-raivdcpmtrsc [Jan 26 10:51:02] DEBUG[1167][C-00000005] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:51:02] DEBUG[1167][C-00000005] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:51:02] DEBUG[1167][C-00000005] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:51:02] VERBOSE[1167][C-00000005] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:51:02] DEBUG[1167][C-00000005] chan_sip.c: Setting SIP_ALREADYGONE on dialog 54c60e01a739-raivdcpmtrsc [Jan 26 10:51:02] DEBUG[1167][C-00000005] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fb25c019bf8' [Jan 26 10:51:02] DEBUG[1167][C-00000005] chan_sip.c: Session timer stopped: 93 - 54c60e01a739-raivdcpmtrsc [Jan 26 10:51:02] VERBOSE[1167][C-00000005] chan_sip.c: Scheduling destruction of SIP dialog '54c60e01a739-raivdcpmtrsc' in 6400 ms (Method: BYE) [Jan 26 10:51:02] DEBUG[1167][C-00000005] chan_sip.c: Received bye, issuing owner hangup [Jan 26 10:51:02] VERBOSE[1167][C-00000005] chan_sip.c: ˙<--- Transmitting (NAT) to #PhonesPublicIP#:2048 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-vh87o96ge8ku;received=#PhonesPublicIP#;rport=2048 ˙From: "799 asterisk11" ;tag=92mfo6km16 ˙To: ;tag=as7d222e17 ˙Call-ID: 54c60e01a739-raivdcpmtrsc ˙CSeq: 3 BYE ˙Server: RSU Telephone Server ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Length: 0 ˙ ˙ ˙<------------> [Jan 26 10:51:02] DEBUG[1167][C-00000005] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:51:02] DEBUG[1490][C-00000005] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 10:51:02] DEBUG[1490][C-00000005] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 10:51:02] DEBUG[1490][C-00000005] channel.c: Set channel SIP/799-00000002 to write format g722 [Jan 26 10:51:02] DEBUG[1490][C-00000005] pbx.c: Spawn extension (localsets-common,200,16) exited non-zero on 'SIP/799-00000002' [Jan 26 10:51:02] VERBOSE[1490][C-00000005] pbx.c: == Spawn extension (localsets-common, 200, 16) exited non-zero on 'SIP/799-00000002' [Jan 26 10:51:02] DEBUG[1490][C-00000005] channel.c: Soft-Hanging up channel 'SIP/799-00000002' [Jan 26 10:51:02] DEBUG[1490][C-00000005] channel.c: Hanging up channel 'SIP/799-00000002' [Jan 26 10:51:02] DEBUG[1490][C-00000005] chan_sip.c: Hangup call SIP/799-00000002, SIP callid 54c60e01a739-raivdcpmtrsc [Jan 26 10:51:02] DEBUG[1490][C-00000005] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fb25c019bf8' [Jan 26 10:51:02] DEBUG[1490][C-00000005] cdr_radius.c: Unable to create RADIUS record. CDR not recorded! [Jan 26 10:51:02] DEBUG[1490][C-00000005] res_config_sqlite.c: SQL query: INSERT INTO ast_cdr (clid,src,dst,dcontext,channel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid) VALUES ('"799 asterisk11" <799>','799','200','localsets-common','SIP/799-00000002','Playback','custom/system-number-is-not-in-service','2015-01-26 10:50:59','2015-01-26 10:50:59','2015-01-26 10:51:02','3','3','ANSWERED','DOCUMENTATION','1422265859.2') [Jan 26 10:51:02] DEBUG[1008] devicestate.c: No provider found, checking channel drivers for SIP - 799 [Jan 26 10:51:02] DEBUG[1008] chan_sip.c: Checking device state for peer 799 [Jan 26 10:51:02] DEBUG[1008] devicestate.c: Changing state for SIP/799 - state 1 (Not in use) [Jan 26 10:51:02] DEBUG[1008] devicestate.c: device 'SIP/799' state '1' [Jan 26 10:51:02] DEBUG[1196] app_queue.c: Device 'SIP/799' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 26 10:51:09] DEBUG[1167] chan_sip.c: Auto destroying SIP dialog '54c60e01a739-raivdcpmtrsc' [Jan 26 10:51:09] DEBUG[1167] chan_sip.c: Destroying SIP dialog 54c60e01a739-raivdcpmtrsc [Jan 26 10:51:09] VERBOSE[1167] chan_sip.c: Really destroying SIP dialog '54c60e01a739-raivdcpmtrsc' Method: BYE [Jan 26 10:51:09] DEBUG[1167] rtp_engine.c: Destroyed RTP instance '0x7fb25c019bf8'