[Jan 26 10:28:00] Asterisk 13.1.0 built by root @ asterisk on a x86_64 running Linux on 2015-01-09 18:57:29 UTC [Jan 26 10:28:02] VERBOSE[1380] 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-sgd8g94a6u1w;rport From: "Test799" ;tag=lxhovjwifc To: "Test799" Call-ID: 386d43a96f0d-0wsg87j7eu2c CSeq: 261 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: 0 Content-Length: 0 <-------------> [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Header 0 [ 50]: REGISTER sip:#HOSTNAME# SIP/2.0 [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-sgd8g94a6u1w;rport [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Header 2 [ 70]: From: "Test799" ;tag=lxhovjwifc [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Header 3 [ 53]: To: "Test799" [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Header 4 [ 34]: Call-ID: 386d43a96f0d-0wsg87j7eu2c [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Header 5 [ 18]: CSeq: 261 REGISTER [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:28:02] DEBUG[1380] 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 10:28:02] DEBUG[1380] chan_sip.c: Header 8 [ 30]: User-Agent: snom300/8.7.3.25.5 [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Header 10 [ 23]: X-Real-IP: 192.168.1.41 [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Header 12 [ 10]: Expires: 0 [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Jan 26 10:28:02] VERBOSE[1380] chan_sip.c: --- (14 headers 0 lines) --- [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: = Looking for Call ID: 386d43a96f0d-0wsg87j7eu2c (Checking From) --From tag lxhovjwifc --To-tag [Jan 26 10:28:02] DEBUG[1380] acl.c: For destination '#PhonesPublicIP#', our source address is '192.168.122.99'. [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.122.99:5060 [Jan 26 10:28:02] DEBUG[1380] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:28:02] DEBUG[1380] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:28:02] VERBOSE[1380] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Allocating new SIP dialog for 386d43a96f0d-0wsg87j7eu2c - REGISTER (No RTP) [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Initializing initreq for method REGISTER - callid 386d43a96f0d-0wsg87j7eu2c [Jan 26 10:28:02] DEBUG[1380] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:28:02] DEBUG[1380] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:28:02] VERBOSE[1380] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:28:02] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:02] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:02] VERBOSE[1380] 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-sgd8g94a6u1w;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=lxhovjwifc To: "Test799" ;tag=as29bca9cc Call-ID: 386d43a96f0d-0wsg87j7eu2c CSeq: 261 REGISTER Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3648165e" Content-Length: 0 <------------> [Jan 26 10:28:02] DEBUG[1380] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:02] VERBOSE[1380] chan_sip.c: Scheduling destruction of SIP dialog '386d43a96f0d-0wsg87j7eu2c' in 32000 ms (Method: REGISTER) [Jan 26 10:28:03] DEBUG[1491] threadpool.c: Worker thread idle timeout reached. Dying. [Jan 26 10:28:03] DEBUG[1339] threadpool.c: Destroying worker thread 8 [Jan 26 10:28:03] VERBOSE[1380] 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-dctnyqarmsgd;rport From: "Test799" ;tag=lxhovjwifc To: "Test799" Call-ID: 386d43a96f0d-0wsg87j7eu2c CSeq: 262 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="asterisk",nonce="3648165e",uri="sip:#HOSTNAME#",response="7674e32a7c5f395286318a0f32e78ea2",algorithm=MD5 Expires: 0 Content-Length: 0 <-------------> [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 0 [ 50]: REGISTER sip:#HOSTNAME# SIP/2.0 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-dctnyqarmsgd;rport [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 2 [ 70]: From: "Test799" ;tag=lxhovjwifc [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 3 [ 53]: To: "Test799" [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 4 [ 34]: Call-ID: 386d43a96f0d-0wsg87j7eu2c [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 5 [ 18]: CSeq: 262 REGISTER [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:28:03] DEBUG[1380] 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 10:28:03] DEBUG[1380] chan_sip.c: Header 8 [ 30]: User-Agent: snom300/8.7.3.25.5 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 10 [ 23]: X-Real-IP: 192.168.1.41 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 12 [168]: Authorization: Digest username="799",realm="asterisk",nonce="3648165e",uri="sip:#HOSTNAME#",response="7674e32a7c5f395286318a0f32e78ea2",algorithm=MD5 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 13 [ 10]: Expires: 0 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: --- (15 headers 0 lines) --- [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: = Looking for Call ID: 386d43a96f0d-0wsg87j7eu2c (Checking From) --From tag lxhovjwifc --To-tag [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Initializing initreq for method REGISTER - callid 386d43a96f0d-0wsg87j7eu2c [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:03] NOTICE[1380] chan_sip.c: Correct auth, but based on stale nonce received from '"Test799" ;tag=lxhovjwifc' [Jan 26 10:28:03] VERBOSE[1380] 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-dctnyqarmsgd;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=lxhovjwifc To: "Test799" ;tag=as781faa42 Call-ID: 386d43a96f0d-0wsg87j7eu2c CSeq: 262 REGISTER Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="38f65525", stale=true Content-Length: 0 <------------> [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: Scheduling destruction of SIP dialog '386d43a96f0d-0wsg87j7eu2c' in 32000 ms (Method: REGISTER) [Jan 26 10:28:03] VERBOSE[1380] 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-cr5472w7c022;rport From: "Test799" ;tag=lxhovjwifc To: "Test799" Call-ID: 386d43a96f0d-0wsg87j7eu2c CSeq: 263 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="asterisk",nonce="38f65525",uri="sip:#HOSTNAME#",response="49e1bd1acf41977be694f978c58f4466",algorithm=MD5 Expires: 0 Content-Length: 0 <-------------> [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 0 [ 50]: REGISTER sip:#HOSTNAME# SIP/2.0 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-cr5472w7c022;rport [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 2 [ 70]: From: "Test799" ;tag=lxhovjwifc [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 3 [ 53]: To: "Test799" [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 4 [ 34]: Call-ID: 386d43a96f0d-0wsg87j7eu2c [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 5 [ 18]: CSeq: 263 REGISTER [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:28:03] DEBUG[1380] 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 10:28:03] DEBUG[1380] chan_sip.c: Header 8 [ 30]: User-Agent: snom300/8.7.3.25.5 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 10 [ 23]: X-Real-IP: 192.168.1.41 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 12 [168]: Authorization: Digest username="799",realm="asterisk",nonce="38f65525",uri="sip:#HOSTNAME#",response="49e1bd1acf41977be694f978c58f4466",algorithm=MD5 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 13 [ 10]: Expires: 0 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: --- (15 headers 0 lines) --- [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: = Looking for Call ID: 386d43a96f0d-0wsg87j7eu2c (Checking From) --From tag lxhovjwifc --To-tag [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Initializing initreq for method REGISTER - callid 386d43a96f0d-0wsg87j7eu2c [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:03] DEBUG[1339] threadpool.c: Increasing threadpool stasis-core's size by 1 [Jan 26 10:28:03] DEBUG[1351] devicestate.c: No provider found, checking channel drivers for SIP - 799 [Jan 26 10:28:03] DEBUG[1351] chan_sip.c: Checking device state for peer 799 [Jan 26 10:28:03] DEBUG[1351] devicestate.c: Changing state for SIP/799 - state 5 (Unavailable) [Jan 26 10:28:03] DEBUG[1394] app_queue.c: Device 'SIP/799' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue. [Jan 26 10:28:03] VERBOSE[1380] 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-cr5472w7c022;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=lxhovjwifc To: "Test799" ;tag=as781faa42 Call-ID: 386d43a96f0d-0wsg87j7eu2c CSeq: 263 REGISTER Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 0 Date: Mon, 26 Jan 2015 09:28:03 GMT Content-Length: 0 <------------> [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: Scheduling destruction of SIP dialog '386d43a96f0d-0wsg87j7eu2c' in 32000 ms (Method: REGISTER) [Jan 26 10:28:03] DEBUG[1351] devicestate.c: No provider found, checking channel drivers for SIP - 799 [Jan 26 10:28:03] DEBUG[1351] chan_sip.c: Checking device state for peer 799 [Jan 26 10:28:03] DEBUG[1351] devicestate.c: Changing state for SIP/799 - state 5 (Unavailable) [Jan 26 10:28:03] VERBOSE[1380] 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-jo5nf8jmk386;rport From: "Test799" ;tag=76ho82lzgv To: "Test799" Call-ID: 386d43a96f0d-0wsg87j7eu2c CSeq: 264 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 10:28:03] DEBUG[1380] chan_sip.c: Header 0 [ 50]: REGISTER sip:#HOSTNAME# SIP/2.0 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-jo5nf8jmk386;rport [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 2 [ 70]: From: "Test799" ;tag=76ho82lzgv [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 3 [ 53]: To: "Test799" [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 4 [ 34]: Call-ID: 386d43a96f0d-0wsg87j7eu2c [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 5 [ 18]: CSeq: 264 REGISTER [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:28:03] DEBUG[1380] 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 10:28:03] DEBUG[1380] chan_sip.c: Header 8 [ 30]: User-Agent: snom300/8.7.3.25.5 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 10 [ 23]: X-Real-IP: 192.168.1.41 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 12 [ 13]: Expires: 3600 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: --- (14 headers 0 lines) --- [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: = Looking for Call ID: 386d43a96f0d-0wsg87j7eu2c (Checking From) --From tag 76ho82lzgv --To-tag [Jan 26 10:28:03] DEBUG[1380] acl.c: For destination '#PhonesPublicIP#', our source address is '192.168.122.99'. [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.122.99:5060 [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Allocating new SIP dialog for 386d43a96f0d-0wsg87j7eu2c - REGISTER (No RTP) [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Initializing initreq for method REGISTER - callid 386d43a96f0d-0wsg87j7eu2c [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:03] VERBOSE[1380] 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-jo5nf8jmk386;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=76ho82lzgv To: "Test799" ;tag=as5bac194c Call-ID: 386d43a96f0d-0wsg87j7eu2c CSeq: 264 REGISTER Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="42cd26c4" Content-Length: 0 <------------> [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: Scheduling destruction of SIP dialog '386d43a96f0d-0wsg87j7eu2c' in 32000 ms (Method: REGISTER) [Jan 26 10:28:03] VERBOSE[1380] 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-jhx0x08nnnvq;rport From: "Test799" ;tag=76ho82lzgv To: "Test799" Call-ID: 386d43a96f0d-0wsg87j7eu2c CSeq: 265 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="asterisk",nonce="42cd26c4",uri="sip:#HOSTNAME#",response="14429bce2a24c42f14782feba7f3927a",algorithm=MD5 Expires: 3600 Content-Length: 0 <-------------> [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 0 [ 50]: REGISTER sip:#HOSTNAME# SIP/2.0 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-jhx0x08nnnvq;rport [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 2 [ 70]: From: "Test799" ;tag=76ho82lzgv [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 3 [ 53]: To: "Test799" [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 4 [ 34]: Call-ID: 386d43a96f0d-0wsg87j7eu2c [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 5 [ 18]: CSeq: 265 REGISTER [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:28:03] DEBUG[1380] 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 10:28:03] DEBUG[1380] chan_sip.c: Header 8 [ 30]: User-Agent: snom300/8.7.3.25.5 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 10 [ 23]: X-Real-IP: 192.168.1.41 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 12 [168]: Authorization: Digest username="799",realm="asterisk",nonce="42cd26c4",uri="sip:#HOSTNAME#",response="14429bce2a24c42f14782feba7f3927a",algorithm=MD5 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: --- (15 headers 0 lines) --- [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: = Looking for Call ID: 386d43a96f0d-0wsg87j7eu2c (Checking From) --From tag 76ho82lzgv --To-tag [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Initializing initreq for method REGISTER - callid 386d43a96f0d-0wsg87j7eu2c [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:03] NOTICE[1380] chan_sip.c: Correct auth, but based on stale nonce received from '"Test799" ;tag=76ho82lzgv' [Jan 26 10:28:03] VERBOSE[1380] 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-jhx0x08nnnvq;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=76ho82lzgv To: "Test799" ;tag=as781faa42 Call-ID: 386d43a96f0d-0wsg87j7eu2c CSeq: 265 REGISTER Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="15f2fa27", stale=true Content-Length: 0 <------------> [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: Scheduling destruction of SIP dialog '386d43a96f0d-0wsg87j7eu2c' in 32000 ms (Method: REGISTER) [Jan 26 10:28:03] VERBOSE[1380] 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-dm5wxect32mf;rport From: "Test799" ;tag=76ho82lzgv To: "Test799" Call-ID: 386d43a96f0d-0wsg87j7eu2c CSeq: 266 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="asterisk",nonce="15f2fa27",uri="sip:#HOSTNAME#",response="a9577f53bdce0d9f019b789ee19ea2c9",algorithm=MD5 Expires: 3600 Content-Length: 0 <-------------> [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 0 [ 50]: REGISTER sip:#HOSTNAME# SIP/2.0 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-dm5wxect32mf;rport [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 2 [ 70]: From: "Test799" ;tag=76ho82lzgv [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 3 [ 53]: To: "Test799" [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 4 [ 34]: Call-ID: 386d43a96f0d-0wsg87j7eu2c [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 5 [ 18]: CSeq: 266 REGISTER [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:28:03] DEBUG[1380] 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 10:28:03] DEBUG[1380] chan_sip.c: Header 8 [ 30]: User-Agent: snom300/8.7.3.25.5 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 9 [ 20]: Allow-Events: dialog [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 10 [ 23]: X-Real-IP: 192.168.1.41 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 11 [ 21]: Supported: path, gruu [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 12 [168]: Authorization: Digest username="799",realm="asterisk",nonce="15f2fa27",uri="sip:#HOSTNAME#",response="a9577f53bdce0d9f019b789ee19ea2c9",algorithm=MD5 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: --- (15 headers 0 lines) --- [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: = Looking for Call ID: 386d43a96f0d-0wsg87j7eu2c (Checking From) --From tag 76ho82lzgv --To-tag [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Initializing initreq for method REGISTER - callid 386d43a96f0d-0wsg87j7eu2c [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:28:03] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:03] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: build_path: do not use Path headers [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Allocating new SIP dialog for 7d06a2fa407e8b9d4431350a0ea36c09@127.0.1.1:5060 - OPTIONS (No RTP) [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jan 26 10:28:03] DEBUG[1380] acl.c: For destination '#PhonesPublicIP#', our source address is '192.168.122.99'. [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.122.99:5060 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: SIP call-id changed from '7d06a2fa407e8b9d4431350a0ea36c09@127.0.1.1:5060' to '3255e085530490e50f98c1a4224986a9@192.168.122.99:5060' [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Initializing initreq for method OPTIONS - callid 3255e085530490e50f98c1a4224986a9@192.168.122.99:5060 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 0 [ 55]: OPTIONS sip:799@192.168.1.41:2048;line=xz99z7xg SIP/2.0 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.122.99:5060;branch=z9hG4bK4d5cd694;rport [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as5547539e [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 4 [ 45]: To: [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 5 [ 43]: Contact: [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 6 [ 61]: Call-ID: 3255e085530490e50f98c1a4224986a9@192.168.122.99:5060 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 8 [ 32]: User-Agent: RSU Telephone Server [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 9 [ 35]: Date: Mon, 26 Jan 2015 09:28:03 GMT [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: Reliably Transmitting (NAT) to #PhonesPublicIP#:2048: OPTIONS sip:799@192.168.1.41:2048;line=xz99z7xg SIP/2.0 Via: SIP/2.0/UDP 192.168.122.99:5060;branch=z9hG4bK4d5cd694;rport Max-Forwards: 70 From: "asterisk" ;tag=as5547539e To: Contact: Call-ID: 3255e085530490e50f98c1a4224986a9@192.168.122.99:5060 CSeq: 102 OPTIONS User-Agent: RSU Telephone Server Date: Mon, 26 Jan 2015 09:28:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #131 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:03] VERBOSE[1380] 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-dm5wxect32mf;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=76ho82lzgv To: "Test799" ;tag=as781faa42 Call-ID: 386d43a96f0d-0wsg87j7eu2c CSeq: 266 REGISTER Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: ;expires=3600 Date: Mon, 26 Jan 2015 09:28:03 GMT Content-Length: 0 <------------> [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:03] DEBUG[1351] devicestate.c: No provider found, checking channel drivers for SIP - 799 [Jan 26 10:28:03] DEBUG[1351] chan_sip.c: Checking device state for peer 799 [Jan 26 10:28:03] DEBUG[1351] devicestate.c: Changing state for SIP/799 - state 1 (Not in use) [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: Scheduling destruction of SIP dialog '386d43a96f0d-0wsg87j7eu2c' in 32000 ms (Method: REGISTER) [Jan 26 10:28:03] DEBUG[1394] 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 10:28:03] VERBOSE[1380] chan_sip.c: <--- SIP read from UDP:#PhonesPublicIP#:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.122.99:5060;branch=z9hG4bK4d5cd694;rport=5060;received=#ServersPublicIP# From: "asterisk" ;tag=as5547539e To: ;tag=ahtg46pcol Call-ID: 3255e085530490e50f98c1a4224986a9@192.168.122.99:5060 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 10:28:03] DEBUG[1380] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 192.168.122.99:5060;branch=z9hG4bK4d5cd694;rport=5060;received=#ServersPublicIP# [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as5547539e [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 3 [ 60]: To: ;tag=ahtg46pcol [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 4 [ 61]: Call-ID: 3255e085530490e50f98c1a4224986a9@192.168.122.99:5060 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 6 [ 59]: Contact: ;reg-id=1 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 7 [ 30]: User-Agent: snom300/8.7.3.25.5 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 10 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: --- (14 headers 0 lines) --- [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: = Looking for Call ID: 3255e085530490e50f98c1a4224986a9@192.168.122.99:5060 (Checking To) --From tag as5547539e --To-tag ahtg46pcol [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #131 [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Stopping retransmission on '3255e085530490e50f98c1a4224986a9@192.168.122.99:5060' of Request 102: Match Found [Jan 26 10:28:03] DEBUG[1380] chan_sip.c: Destroying SIP dialog 3255e085530490e50f98c1a4224986a9@192.168.122.99:5060 [Jan 26 10:28:03] VERBOSE[1380] chan_sip.c: Really destroying SIP dialog '3255e085530490e50f98c1a4224986a9@192.168.122.99:5060' Method: OPTIONS [Jan 26 10:28:09] VERBOSE[1380] 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-nzacef7o3iiw;rport From: "Test799" ;tag=ssrqzr5rru To: Call-ID: 54c608a80571-kfzkg8mngogd 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: 485 v=0 o=root 369527818 369527818 IN IP4 192.168.1.41 s=call c=IN IP4 192.168.1.41 t=0 0 m=audio 62770 RTP/AVP 9 0 8 3 99 108 18 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:t26a9YKjs29YZJCQk8eIc9JcZkWQbq3+vT3+Xgq2 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 10:28:09] DEBUG[1380] chan_sip.c: Header 0 [ 63]: INVITE sip:200@#HOSTNAME#;user=phone SIP/2.0 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-nzacef7o3iiw;rport [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 2 [ 70]: From: "Test799" ;tag=ssrqzr5rru [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 3 [ 54]: To: [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 4 [ 34]: Call-ID: 54c608a80571-kfzkg8mngogd [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 7 [ 59]: Contact: ;reg-id=1 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 8 [ 28]: X-Serialnumber: 0004133BD3D8 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 9 [ 21]: P-Key-Flags: keys="3" [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 10 [ 30]: User-Agent: snom300/8.7.3.25.5 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 11 [ 23]: Accept: application/sdp [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 12 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 13 [ 42]: Allow-Events: talk, hold, refer, call-info [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 14 [ 47]: Supported: timer, 100rel, replaces, from-change [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 15 [ 35]: Session-Expires: 3600;refresher=uas [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 16 [ 10]: Min-SE: 90 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 17 [ 29]: Content-Type: application/sdp [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 18 [ 19]: Content-Length: 485 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 19 [ 0]: [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 0 [ 3]: v=0 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 1 [ 46]: o=root 369527818 369527818 IN IP4 192.168.1.41 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 2 [ 6]: s=call [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.41 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 5 [ 43]: m=audio 62770 RTP/AVP 9 0 8 3 99 108 18 101 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:t26a9YKjs29YZJCQk8eIc9JcZkWQbq3+vT3+Xgq2 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 7 [ 20]: a=rtpmap:9 G722/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 10 [ 19]: a=rtpmap:3 GSM/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 11 [ 24]: a=rtpmap:99 G726-32/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 12 [ 30]: a=rtpmap:108 AAL2-G726-32/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 13 [ 21]: a=rtpmap:18 G729/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 14 [ 19]: a=fmtp:18 annexb=no [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 15 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 16 [ 15]: a=fmtp:101 0-15 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 17 [ 10]: a=ptime:20 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 18 [ 10]: a=sendrecv [Jan 26 10:28:09] VERBOSE[1380] chan_sip.c: --- (19 headers 19 lines) --- [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: = Looking for Call ID: 54c608a80571-kfzkg8mngogd (Checking From) --From tag ssrqzr5rru --To-tag [Jan 26 10:28:09] DEBUG[1380] acl.c: For destination '#PhonesPublicIP#', our source address is '192.168.122.99'. [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.122.99:5060 [Jan 26 10:28:09] DEBUG[1380] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:28:09] DEBUG[1380] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:28:09] VERBOSE[1380] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Allocating new SIP dialog for 54c608a80571-kfzkg8mngogd - INVITE (No RTP) [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 26 10:28:09] DEBUG[1380][C-00000003] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change" [Jan 26 10:28:09] DEBUG[1380][C-00000003] sip/reqresp_parser.c: Found SIP option: -timer- [Jan 26 10:28:09] DEBUG[1380][C-00000003] sip/reqresp_parser.c: Matched SIP option: timer [Jan 26 10:28:09] DEBUG[1380][C-00000003] sip/reqresp_parser.c: Found SIP option: -100rel- [Jan 26 10:28:09] DEBUG[1380][C-00000003] sip/reqresp_parser.c: Matched SIP option: 100rel [Jan 26 10:28:09] DEBUG[1380][C-00000003] sip/reqresp_parser.c: Found SIP option: -replaces- [Jan 26 10:28:09] DEBUG[1380][C-00000003] sip/reqresp_parser.c: Matched SIP option: replaces [Jan 26 10:28:09] DEBUG[1380][C-00000003] sip/reqresp_parser.c: Found SIP option: -from-change- [Jan 26 10:28:09] DEBUG[1380][C-00000003] sip/reqresp_parser.c: Matched SIP option: from-change [Jan 26 10:28:09] DEBUG[1380][C-00000003] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:28:09] DEBUG[1380][C-00000003] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Initializing initreq for method INVITE - callid 54c608a80571-kfzkg8mngogd [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Using INVITE request as basis request - 54c608a80571-kfzkg8mngogd [Jan 26 10:28:09] DEBUG[1380][C-00000003] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:09] DEBUG[1380][C-00000003] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found peer '799' for '799' from #PhonesPublicIP#:2048 [Jan 26 10:28:09] VERBOSE[1380][C-00000003] 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-nzacef7o3iiw;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=ssrqzr5rru To: ;tag=as200b7970 Call-ID: 54c608a80571-kfzkg8mngogd CSeq: 1 INVITE Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="333db908" Content-Length: 0 <------------> [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #135 [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Scheduling destruction of SIP dialog '54c608a80571-kfzkg8mngogd' in 6400 ms (Method: INVITE) [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: SIP TIMER: Rescheduling retransmission #135 (1) SIP/2.0 - 1 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #135)) [Jan 26 10:28:09] VERBOSE[1380] 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-nzacef7o3iiw;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=ssrqzr5rru To: ;tag=as200b7970 Call-ID: 54c608a80571-kfzkg8mngogd CSeq: 1 INVITE Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="333db908" Content-Length: 0 --- [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:09] VERBOSE[1380] 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-nzacef7o3iiw;rport From: "Test799" ;tag=ssrqzr5rru To: ;tag=as200b7970 Call-ID: 54c608a80571-kfzkg8mngogd CSeq: 1 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 0 [ 60]: ACK sip:200@#HOSTNAME#;user=phone SIP/2.0 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-nzacef7o3iiw;rport [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 2 [ 70]: From: "Test799" ;tag=ssrqzr5rru [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 3 [ 69]: To: ;tag=as200b7970 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 4 [ 34]: Call-ID: 54c608a80571-kfzkg8mngogd [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 7 [ 59]: Contact: ;reg-id=1 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 26 10:28:09] VERBOSE[1380] chan_sip.c: --- (9 headers 0 lines) --- [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: = Looking for Call ID: 54c608a80571-kfzkg8mngogd (Checking From) --From tag ssrqzr5rru --To-tag as200b7970 [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #135 [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Stopping retransmission on '54c608a80571-kfzkg8mngogd' of Response 1: Match Found [Jan 26 10:28:09] VERBOSE[1380] 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-3aapd8nrw8yv;rport From: "Test799" ;tag=ssrqzr5rru To: Call-ID: 54c608a80571-kfzkg8mngogd 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="asterisk",nonce="333db908",uri="sip:200@#HOSTNAME#;user=phone",response="921c2e77ad5ad614681ddba9fb77dc5c",algorithm=MD5 Content-Type: application/sdp Content-Length: 485 v=0 o=root 369527818 369527818 IN IP4 192.168.1.41 s=call c=IN IP4 192.168.1.41 t=0 0 m=audio 62770 RTP/AVP 9 0 8 3 99 108 18 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:t26a9YKjs29YZJCQk8eIc9JcZkWQbq3+vT3+Xgq2 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 10:28:09] DEBUG[1380] chan_sip.c: Header 0 [ 63]: INVITE sip:200@#HOSTNAME#;user=phone SIP/2.0 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-3aapd8nrw8yv;rport [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 2 [ 70]: From: "Test799" ;tag=ssrqzr5rru [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 3 [ 54]: To: [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 4 [ 34]: Call-ID: 54c608a80571-kfzkg8mngogd [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 5 [ 14]: CSeq: 2 INVITE [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 7 [ 59]: Contact: ;reg-id=1 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 8 [ 28]: X-Serialnumber: 0004133BD3D8 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 9 [ 21]: P-Key-Flags: keys="3" [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 10 [ 30]: User-Agent: snom300/8.7.3.25.5 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 11 [ 23]: Accept: application/sdp [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 12 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 13 [ 42]: Allow-Events: talk, hold, refer, call-info [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 14 [ 47]: Supported: timer, 100rel, replaces, from-change [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 15 [ 35]: Session-Expires: 3600;refresher=uas [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 16 [ 10]: Min-SE: 90 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 17 [183]: Authorization: Digest username="799",realm="asterisk",nonce="333db908",uri="sip:200@#HOSTNAME#;user=phone",response="921c2e77ad5ad614681ddba9fb77dc5c",algorithm=MD5 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 18 [ 29]: Content-Type: application/sdp [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 19 [ 19]: Content-Length: 485 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 20 [ 0]: [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 0 [ 3]: v=0 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 1 [ 46]: o=root 369527818 369527818 IN IP4 192.168.1.41 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 2 [ 6]: s=call [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.41 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 5 [ 43]: m=audio 62770 RTP/AVP 9 0 8 3 99 108 18 101 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:t26a9YKjs29YZJCQk8eIc9JcZkWQbq3+vT3+Xgq2 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 7 [ 20]: a=rtpmap:9 G722/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 10 [ 19]: a=rtpmap:3 GSM/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 11 [ 24]: a=rtpmap:99 G726-32/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 12 [ 30]: a=rtpmap:108 AAL2-G726-32/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 13 [ 21]: a=rtpmap:18 G729/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 14 [ 19]: a=fmtp:18 annexb=no [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 15 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 16 [ 15]: a=fmtp:101 0-15 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 17 [ 10]: a=ptime:20 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Body 18 [ 10]: a=sendrecv [Jan 26 10:28:09] VERBOSE[1380] chan_sip.c: --- (20 headers 19 lines) --- [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: = Looking for Call ID: 54c608a80571-kfzkg8mngogd (Checking From) --From tag ssrqzr5rru --To-tag [Jan 26 10:28:09] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:09] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:09] DEBUG[1380] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:09] DEBUG[1380] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 26 10:28:09] DEBUG[1380][C-00000003] netsock2.c: Splitting '192.168.1.41:2048' into... [Jan 26 10:28:09] DEBUG[1380][C-00000003] netsock2.c: ...host '192.168.1.41' and port '2048'. [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: NAT detected for 192.168.1.41 / #PhonesPublicIP# [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Sending to #PhonesPublicIP#:2048 (NAT) [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Initializing initreq for method INVITE - callid 54c608a80571-kfzkg8mngogd [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Using INVITE request as basis request - 54c608a80571-kfzkg8mngogd [Jan 26 10:28:09] DEBUG[1380][C-00000003] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:09] DEBUG[1380][C-00000003] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found peer '799' for '799' from #PhonesPublicIP#:2048 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7fef9400c4d8' [Jan 26 10:28:09] DEBUG[1380][C-00000003] res_rtp_asterisk.c: Allocated port 16188 for RTP instance '0x7fef9400c4d8' [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: RTP instance '0x7fef9400c4d8' is setup and ready to go [Jan 26 10:28:09] DEBUG[1380][C-00000003] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7fef9400c4d8' [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Setting NAT on RTP to On [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing session-level SDP o=root 369527818 369527818 IN IP4 192.168.1.41... OK. [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED OR FAILED. [Jan 26 10:28:09] DEBUG[1380][C-00000003] netsock2.c: Splitting '192.168.1.41' into... [Jan 26 10:28:09] DEBUG[1380][C-00000003] netsock2.c: ...host '192.168.1.41' and port ''. [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.41... OK. [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found RTP audio format 9 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Setting payload 9 (0x7fef9401d3b8) based on m type on 0x7fefd3934320 [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found RTP audio format 0 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Setting payload 0 (0x7fef9400cb78) based on m type on 0x7fefd3934320 [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found RTP audio format 8 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Setting payload 8 (0x7fef94010238) based on m type on 0x7fefd3934320 [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found RTP audio format 3 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Setting payload 3 (0x7fef94014198) based on m type on 0x7fefd3934320 [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found RTP audio format 99 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Setting payload 99 (0x7fef940149d8) based on m type on 0x7fefd3934320 [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found RTP audio format 108 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Setting payload 108 (0x7fef94015218) based on m type on 0x7fefd3934320 [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found RTP audio format 18 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Setting payload 18 (0x7fef94015678) based on m type on 0x7fefd3934320 [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found RTP audio format 101 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Setting payload 101 (0x7fef9401e418) based on m type on 0x7fefd3934320 [Jan 26 10:28:09] DEBUG[1380][C-00000003] sdp_srtp.c: No SRTP module loaded, can't setup SRTP session. [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:t26a9YKjs29YZJCQk8eIc9JcZkWQbq3+vT3+Xgq2... UNSUPPORTED OR FAILED. [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found audio description format G722 for ID 9 [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found audio description format GSM for ID 3 [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK. [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found audio description format G726-32 for ID 99 [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:99 G726-32/8000... OK. [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found audio description format AAL2-G726-32 for ID 108 [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:108 AAL2-G726-32/8000... OK. [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found audio description format G729 for ID 18 [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... OK. [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Capabilities: us - (g722|alaw|ulaw), peer - audio=(ulaw|gsm|alaw|g722|g729|g726|g726aal2)/video=(nothing)/text=(nothing), combined - (g722|alaw|ulaw) [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 26 10:28:09] DEBUG[1380][C-00000003] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fef9400c4d8' [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Peer audio RTP is at port 192.168.1.41:62770 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Copying payload 0 (0x7fef9401d3b8) from 0x7fefd3934320 to 0x7fef9400c618 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Copying payload 3 (0x7fef94010238) from 0x7fefd3934320 to 0x7fef9400c618 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Copying payload 8 (0x7fef9400cb78) from 0x7fefd3934320 to 0x7fef9400c618 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Copying payload 9 (0x7fef9400cbf8) from 0x7fefd3934320 to 0x7fef9400c618 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Copying payload 18 (0x7fef94015218) from 0x7fefd3934320 to 0x7fef9400c618 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Copying payload 99 (0x7fef94014198) from 0x7fefd3934320 to 0x7fef9400c618 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Copying payload 101 (0x7fef94015678) from 0x7fefd3934320 to 0x7fef9400c618 [Jan 26 10:28:09] DEBUG[1380][C-00000003] rtp_engine.c: Copying payload 108 (0x7fef940149d8) from 0x7fefd3934320 to 0x7fef9400c618 [Jan 26 10:28:09] DEBUG[1380][C-00000003] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7fef9400c4d8' [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: We're settling with these formats: (g722|alaw|ulaw) [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Checking SIP call limits for device 799 [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Updating call counter for incoming call [Jan 26 10:28:09] DEBUG[1380][C-00000003] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:09] DEBUG[1380][C-00000003] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:09] DEBUG[1380][C-00000003] netsock2.c: Splitting '#HOSTNAME#' into... [Jan 26 10:28:09] DEBUG[1380][C-00000003] netsock2.c: ...host '#HOSTNAME#' and port ''. [Jan 26 10:28:09] VERBOSE[1380][C-00000003] chan_sip.c: Looking for 200 in localsets-common (domain #HOSTNAME#) [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Incoming INVITE with 'timer' option supported [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: INVITE also has "Session-Expires" header. [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Session-Expires: 3600 [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Refresher: UAS [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: INVITE also has "Min-SE" header. [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Received Min-SE: 90 [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: *** Our native formats are (g722) [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: *** Joint capabilities are (g722|alaw|ulaw) [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: *** Our capabilities are (g722|alaw|ulaw) [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: *** AST_CODEC_CHOOSE formats are g722 [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: This channel will not be able to handle video. [Jan 26 10:28:09] VERBOSE[1380][C-00000003] sip/route.c: sip_route_dump: route/path hop: [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: SIP/799-00000002: New call is still down.... Trying... [Jan 26 10:28:09] VERBOSE[1380][C-00000003] 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-3aapd8nrw8yv;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=ssrqzr5rru To: Call-ID: 54c608a80571-kfzkg8mngogd CSeq: 2 INVITE Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Jan 26 10:28:09] DEBUG[1380][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:09] DEBUG[1351] devicestate.c: No provider found, checking channel drivers for SIP - 799 [Jan 26 10:28:09] DEBUG[1351] chan_sip.c: Checking device state for peer 799 [Jan 26 10:28:09] DEBUG[1351] devicestate.c: Changing state for SIP/799 - state 1 (Not in use) [Jan 26 10:28:09] DEBUG[1496][C-00000003] pbx.c: Launching 'SIPAddHeader' [Jan 26 10:28:09] DEBUG[1496][C-00000003] chan_sip.c: SIP Header added "Privacy: user; header; session" as __SIPADDHEADER01 [Jan 26 10:28:09] DEBUG[1496][C-00000003] pbx.c: Launching 'Answer' [Jan 26 10:28:09] DEBUG[1496][C-00000003] chan_sip.c: SIP answering channel: SIP/799-00000002 [Jan 26 10:28:09] DEBUG[1496][C-00000003] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jan 26 10:28:09] DEBUG[1351] devicestate.c: No provider found, checking channel drivers for SIP - 799 [Jan 26 10:28:09] DEBUG[1351] chan_sip.c: Checking device state for peer 799 [Jan 26 10:28:09] DEBUG[1351] devicestate.c: Changing state for SIP/799 - state 1 (Not in use) [Jan 26 10:28:09] DEBUG[1496][C-00000003] chan_sip.c: Setting framing from config on incoming call [Jan 26 10:28:09] DEBUG[1496][C-00000003] chan_sip.c: ** Our capability: (g722|alaw|ulaw) Video flag: True Text flag: True [Jan 26 10:28:09] DEBUG[1496][C-00000003] chan_sip.c: ** Our prefcodec: (nothing) [Jan 26 10:28:09] VERBOSE[1496][C-00000003] chan_sip.c: Audio is at 16188 [Jan 26 10:28:09] VERBOSE[1496][C-00000003] chan_sip.c: Adding codec g722 to SDP [Jan 26 10:28:09] VERBOSE[1496][C-00000003] chan_sip.c: Adding codec alaw to SDP [Jan 26 10:28:09] VERBOSE[1496][C-00000003] chan_sip.c: Adding codec ulaw to SDP [Jan 26 10:28:09] VERBOSE[1496][C-00000003] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 26 10:28:09] DEBUG[1496][C-00000003] chan_sip.c: -- Done with adding codecs to SDP [Jan 26 10:28:09] DEBUG[1496][C-00000003] chan_sip.c: Done building SDP. Settling with this capability: (g722|alaw|ulaw) [Jan 26 10:28:09] VERBOSE[1496][C-00000003] 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-3aapd8nrw8yv;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=ssrqzr5rru To: ;tag=as341cfc6c Call-ID: 54c608a80571-kfzkg8mngogd CSeq: 2 INVITE Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 300 v=0 o=peter-marrat 1043518825 1043518825 IN IP4 192.168.122.99 s=RSU Telephone Server c=IN IP4 192.168.122.99 t=0 0 m=audio 16188 RTP/AVP 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=maxptime:150 a=sendrecv <------------> [Jan 26 10:28:09] DEBUG[1496][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #138 [Jan 26 10:28:09] DEBUG[1496][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:09] DEBUG[1496][C-00000003] chan_sip.c: Session timer started: 139 - 54c608a80571-kfzkg8mngogd 900000ms [Jan 26 10:28:09] VERBOSE[1380] 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-nzacef7o3iiw;rport From: "Test799" ;tag=ssrqzr5rru To: ;tag=as200b7970 Call-ID: 54c608a80571-kfzkg8mngogd CSeq: 1 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 0 [ 60]: ACK sip:200@#HOSTNAME#;user=phone SIP/2.0 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-nzacef7o3iiw;rport [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 2 [ 70]: From: "Test799" ;tag=ssrqzr5rru [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 3 [ 69]: To: ;tag=as200b7970 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 4 [ 34]: Call-ID: 54c608a80571-kfzkg8mngogd [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 7 [ 59]: Contact: ;reg-id=1 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 26 10:28:09] VERBOSE[1380] chan_sip.c: --- (9 headers 0 lines) --- [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: = Looking for Call ID: 54c608a80571-kfzkg8mngogd (Checking From) --From tag ssrqzr5rru --To-tag as200b7970 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Invalid SIP message - rejected , no callid, len 413 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: SIP TIMER: Rescheduling retransmission #138 (1) SIP/2.0 - 1 [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #138)) [Jan 26 10:28:09] VERBOSE[1380] 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-3aapd8nrw8yv;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=ssrqzr5rru To: ;tag=as341cfc6c Call-ID: 54c608a80571-kfzkg8mngogd CSeq: 2 INVITE Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 300 v=0 o=peter-marrat 1043518825 1043518825 IN IP4 192.168.122.99 s=RSU Telephone Server c=IN IP4 192.168.122.99 t=0 0 m=audio 16188 RTP/AVP 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=maxptime:150 a=sendrecv --- [Jan 26 10:28:09] DEBUG[1380] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:10] DEBUG[1380] chan_sip.c: SIP TIMER: Rescheduling retransmission #138 (2) SIP/2.0 - 1 [Jan 26 10:28:10] DEBUG[1380] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #138)) [Jan 26 10:28:10] VERBOSE[1380] chan_sip.c: Retransmitting #2 (NAT) to #PhonesPublicIP#:2048: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-3aapd8nrw8yv;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=ssrqzr5rru To: ;tag=as341cfc6c Call-ID: 54c608a80571-kfzkg8mngogd CSeq: 2 INVITE Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 300 v=0 o=peter-marrat 1043518825 1043518825 IN IP4 192.168.122.99 s=RSU Telephone Server c=IN IP4 192.168.122.99 t=0 0 m=audio 16188 RTP/AVP 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=maxptime:150 a=sendrecv --- [Jan 26 10:28:10] DEBUG[1380] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:10] DEBUG[1496][C-00000003] channel.c: Didn't receive a media frame from SIP/799-00000002 within 500 ms of answering. Continuing anyway [Jan 26 10:28:10] DEBUG[1496][C-00000003] pbx.c: Launching 'SayNumber' [Jan 26 10:28:10] DEBUG[1496][C-00000003] channel.c: Set channel SIP/799-00000002 to write format gsm [Jan 26 10:28:10] DEBUG[1496][C-00000003] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jan 26 10:28:10] DEBUG[1380] chan_sip.c: SIP TIMER: Rescheduling retransmission #138 (3) SIP/2.0 - 1 [Jan 26 10:28:10] DEBUG[1380] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #138)) [Jan 26 10:28:10] VERBOSE[1380] chan_sip.c: Retransmitting #3 (NAT) to #PhonesPublicIP#:2048: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-3aapd8nrw8yv;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=ssrqzr5rru To: ;tag=as341cfc6c Call-ID: 54c608a80571-kfzkg8mngogd CSeq: 2 INVITE Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 300 v=0 o=peter-marrat 1043518825 1043518825 IN IP4 192.168.122.99 s=RSU Telephone Server c=IN IP4 192.168.122.99 t=0 0 m=audio 16188 RTP/AVP 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=maxptime:150 a=sendrecv --- [Jan 26 10:28:10] DEBUG[1380] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:10] DEBUG[1496][C-00000003] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 10:28:10] DEBUG[1496][C-00000003] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 10:28:10] DEBUG[1496][C-00000003] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 10:28:10] DEBUG[1496][C-00000003] channel.c: Set channel SIP/799-00000002 to write format g722 [Jan 26 10:28:10] DEBUG[1496][C-00000003] pbx.c: Launching 'Playback' [Jan 26 10:28:10] DEBUG[1496][C-00000003] channel.c: Set channel SIP/799-00000002 to write format gsm [Jan 26 10:28:10] DEBUG[1496][C-00000003] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jan 26 10:28:11] DEBUG[1380] chan_sip.c: SIP TIMER: Rescheduling retransmission #138 (4) SIP/2.0 - 1 [Jan 26 10:28:11] DEBUG[1380] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #138)) [Jan 26 10:28:11] VERBOSE[1380] chan_sip.c: Retransmitting #4 (NAT) to #PhonesPublicIP#:2048: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-3aapd8nrw8yv;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=ssrqzr5rru To: ;tag=as341cfc6c Call-ID: 54c608a80571-kfzkg8mngogd CSeq: 2 INVITE Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 300 v=0 o=peter-marrat 1043518825 1043518825 IN IP4 192.168.122.99 s=RSU Telephone Server c=IN IP4 192.168.122.99 t=0 0 m=audio 16188 RTP/AVP 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=maxptime:150 a=sendrecv --- [Jan 26 10:28:11] DEBUG[1380] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:12] DEBUG[1496][C-00000003] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 10:28:12] DEBUG[1496][C-00000003] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 10:28:12] DEBUG[1496][C-00000003] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 10:28:12] DEBUG[1496][C-00000003] channel.c: Set channel SIP/799-00000002 to write format g722 [Jan 26 10:28:12] DEBUG[1496][C-00000003] pbx.c: Launching 'Hangup' [Jan 26 10:28:12] DEBUG[1496][C-00000003] channel.c: Soft-Hanging (0x20) up channel 'SIP/799-00000002' [Jan 26 10:28:12] DEBUG[1496][C-00000003] pbx.c: Spawn extension (localsets-common,200,5) exited non-zero on 'SIP/799-00000002' [Jan 26 10:28:12] DEBUG[1496][C-00000003] channel.c: Soft-Hanging (0x10) up channel 'SIP/799-00000002' [Jan 26 10:28:12] DEBUG[1496][C-00000003] channel.c: Hanging up channel 'SIP/799-00000002' [Jan 26 10:28:12] DEBUG[1496][C-00000003] chan_sip.c: Hangup call SIP/799-00000002, SIP callid 54c608a80571-kfzkg8mngogd [Jan 26 10:28:12] DEBUG[1496][C-00000003] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fef9400c4d8' [Jan 26 10:28:12] VERBOSE[1496][C-00000003] chan_sip.c: Scheduling destruction of SIP dialog '54c608a80571-kfzkg8mngogd' in 6400 ms (Method: INVITE) [Jan 26 10:28:12] DEBUG[1496][C-00000003] chan_sip.c: Session timer stopped: 139 - 54c608a80571-kfzkg8mngogd [Jan 26 10:28:12] DEBUG[1352] cdr.c: Finalized CDR for SIP/799-00000002 - start 1422264489.726876 answer 1422264489.729236 end 1422264492.358352 dispo ANSWERED [Jan 26 10:28:12] DEBUG[1351] devicestate.c: No provider found, checking channel drivers for SIP - 799 [Jan 26 10:28:12] DEBUG[1351] chan_sip.c: Checking device state for peer 799 [Jan 26 10:28:12] DEBUG[1351] devicestate.c: Changing state for SIP/799 - state 1 (Not in use) [Jan 26 10:28:12] DEBUG[1380] chan_sip.c: SIP TIMER: Rescheduling retransmission #138 (5) SIP/2.0 - 1 [Jan 26 10:28:12] DEBUG[1380] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #138)) [Jan 26 10:28:12] VERBOSE[1380] chan_sip.c: Retransmitting #5 (NAT) to #PhonesPublicIP#:2048: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-3aapd8nrw8yv;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=ssrqzr5rru To: ;tag=as341cfc6c Call-ID: 54c608a80571-kfzkg8mngogd CSeq: 2 INVITE Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 300 v=0 o=peter-marrat 1043518825 1043518825 IN IP4 192.168.122.99 s=RSU Telephone Server c=IN IP4 192.168.122.99 t=0 0 m=audio 16188 RTP/AVP 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=maxptime:150 a=sendrecv --- [Jan 26 10:28:12] DEBUG[1380] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:15] DEBUG[1380] chan_sip.c: Auto destroying SIP dialog '386d43a96f0d-0wsg87j7eu2c' [Jan 26 10:28:15] DEBUG[1380] chan_sip.c: Destroying SIP dialog 386d43a96f0d-0wsg87j7eu2c [Jan 26 10:28:15] VERBOSE[1380] chan_sip.c: Really destroying SIP dialog '386d43a96f0d-0wsg87j7eu2c' Method: REGISTER [Jan 26 10:28:16] DEBUG[1380] chan_sip.c: SIP TIMER: Rescheduling retransmission #138 (6) SIP/2.0 - 1 [Jan 26 10:28:16] DEBUG[1380] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 100 ms (Retrans id #138)) [Jan 26 10:28:16] VERBOSE[1380] chan_sip.c: Retransmitting #6 (NAT) to #PhonesPublicIP#:2048: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.41:2048;branch=z9hG4bK-3aapd8nrw8yv;received=#PhonesPublicIP#;rport=2048 From: "Test799" ;tag=ssrqzr5rru To: ;tag=as341cfc6c Call-ID: 54c608a80571-kfzkg8mngogd CSeq: 2 INVITE Server: RSU Telephone Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 300 v=0 o=peter-marrat 1043518825 1043518825 IN IP4 192.168.122.99 s=RSU Telephone Server c=IN IP4 192.168.122.99 t=0 0 m=audio 16188 RTP/AVP 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=maxptime:150 a=sendrecv --- [Jan 26 10:28:16] DEBUG[1380] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for #PhonesPublicIP#:2048 [Jan 26 10:28:16] WARNING[1380] chan_sip.c: Retransmission timeout reached on transmission 54c608a80571-kfzkg8mngogd for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6399ms with no response [Jan 26 10:28:16] DEBUG[1380] chan_sip.c: Setting SIP_ALREADYGONE on dialog 54c608a80571-kfzkg8mngogd [Jan 26 10:28:16] DEBUG[1380] chan_sip.c: Destroying SIP dialog 54c608a80571-kfzkg8mngogd [Jan 26 10:28:16] VERBOSE[1380] chan_sip.c: Really destroying SIP dialog '54c608a80571-kfzkg8mngogd' Method: INVITE [Jan 26 10:28:16] DEBUG[1380] rtp_engine.c: Destroyed RTP instance '0x7fef9400c4d8' [Jan 26 10:28:32] DEBUG[1495] threadpool.c: Worker thread idle timeout reached. Dying. [Jan 26 10:28:32] DEBUG[1339] threadpool.c: Destroying worker thread 9 [Jan 26 10:28:34] DEBUG[1380] chan_sip.c: Auto destroying SIP dialog '386d43a96f0d-0wsg87j7eu2c' [Jan 26 10:28:34] DEBUG[1380] chan_sip.c: Destroying SIP dialog 386d43a96f0d-0wsg87j7eu2c [Jan 26 10:28:34] VERBOSE[1380] chan_sip.c: Really destroying SIP dialog '386d43a96f0d-0wsg87j7eu2c' Method: REGISTER [Jan 26 10:28:35] DEBUG[1380] chan_sip.c: Auto destroying SIP dialog '386d43a96f0d-0wsg87j7eu2c' [Jan 26 10:28:35] DEBUG[1380] chan_sip.c: Destroying SIP dialog 386d43a96f0d-0wsg87j7eu2c [Jan 26 10:28:35] VERBOSE[1380] chan_sip.c: Really destroying SIP dialog '386d43a96f0d-0wsg87j7eu2c' Method: REGISTER