SIP Debugging enabled *CLI> [Sep 11 10:07:42] DEBUG[18453]: chan_sip.c:1879 retrans_pkt: SIP TIMER: Not rescheduling id #162:OPTIONS (Method 3) (No timer T1) Retransmitting #3 (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK20d63dbc;rport From: "asterisk" ;tag=as0b4d03b0 To: Contact: Call-ID: 760206a7195d8cda2def44b26a762d3e@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:07:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:07:43] DEBUG[18453]: chan_sip.c:1879 retrans_pkt: SIP TIMER: Not rescheduling id #162:OPTIONS (Method 3) (No timer T1) Retransmitting #4 (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK20d63dbc;rport From: "asterisk" ;tag=as0b4d03b0 To: Contact: Call-ID: 760206a7195d8cda2def44b26a762d3e@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:07:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:07:43] DEBUG[18453]: chan_sip.c:3165 sip_destroy: Destroying SIP dialog 760206a7195d8cda2def44b26a762d3e@pbx.example.net Really destroying SIP dialog '760206a7195d8cda2def44b26a762d3e@pbx.example.net' Method: OPTIONS [Sep 11 10:07:43] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/EiconSoftIP [Sep 11 10:07:43] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - EiconSoftIP [Sep 11 10:07:43] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer EiconSoftIP [Sep 11 10:07:43] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/EiconSoftIP - state 1 (Not in use) [Sep 11 10:07:43] DEBUG[18500]: app_queue.c:548 changethread: Device 'SIP/EiconSoftIP' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.250:5060 ---> INVITE sip:701@pbx.example.net;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-xsso7882mmo3;rport From: "Theo Belder" ;tag=bjxxjlf0hu To: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/6.5.10 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 271 v=0 o=root 490570670 490570670 IN IP4 192.168.161.250 s=call c=IN IP4 192.168.161.250 t=0 0 m=audio 15084 RTP/AVP 8 0 3 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:701@pbx.example.net;user=phone SIP/2.0 (49) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-xsso7882mmo3;rport (71) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Theo Belder" ;tag=bjxxjlf0hu (60) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (40) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC (55) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 1 INVITE (14) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1 (63) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: P-Key-Flags: keys="3" (21) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: User-Agent: snom320/6.5.10 (26) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Accept: application/sdp (23) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: Session-Expires: 3600;refresher=uas (35) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 15: Min-SE: 90 (10) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 16: Content-Type: application/sdp (29) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 17: Content-Length: 271 (19) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 18: (0) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 490570670 490570670 IN IP4 192.168.161.250 (49) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.250 (24) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 15084 RTP/AVP 8 0 3 101 (31) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:0 pcmu/8000 (20) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:3 gsm/8000 (19) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (18 headers 13 lines) --- [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:2624 do_setnat: Setting NAT on RTP to Off [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:2629 do_setnat: Setting NAT on VRTP to Off [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:2634 do_setnat: Setting NAT on UDPTL to Off [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for 3c26823294ed-t2wihiydodry@snom320-0004132496DC - INVITE (With RTP) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:1688 parse_sip_options: Begin: parsing SIP "Supported: timer, 100rel, replaces, callerid" [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:1696 parse_sip_options: Found SIP option: -timer- [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:1702 parse_sip_options: Matched SIP option: timer [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:1696 parse_sip_options: Found SIP option: -100rel- [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:1702 parse_sip_options: Matched SIP option: 100rel [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:1696 parse_sip_options: Found SIP option: -replaces- [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:1702 parse_sip_options: Matched SIP option: replaces [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:1696 parse_sip_options: Found SIP option: -callerid- [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:1710 parse_sip_options: Found no match for SIP option: callerid (Please file bug report!) Sending to 192.168.161.250 : 5060 (NAT) Using INVITE request as basis request - 3c26823294ed-t2wihiydodry@snom320-0004132496DC [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:2624 do_setnat: Setting NAT on RTP to Off [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:2629 do_setnat: Setting NAT on VRTP to Off [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:2634 do_setnat: Setting NAT on UDPTL to Off <--- Reliably Transmitting (no NAT) to 192.168.161.250:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-xsso7882mmo3;received=192.168.161.250;rport=5060 From: "Theo Belder" ;tag=bjxxjlf0hu To: ;tag=as5a7a6c45 Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 1 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="siprealm", nonce="1cdcdc03" Content-Length: 0 <------------> [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #174 Scheduling destruction of SIP dialog '3c26823294ed-t2wihiydodry@snom320-0004132496DC' in 32000 ms (Method: INVITE) Found user '403' <--- SIP read from 192.168.161.250:5060 ---> ACK sip:701@pbx.example.net;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-xsso7882mmo3;rport From: "Theo Belder" ;tag=bjxxjlf0hu To: ;tag=as5a7a6c45 Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: ACK sip:701@pbx.example.net;user=phone SIP/2.0 (46) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-xsso7882mmo3;rport (71) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Theo Belder" ;tag=bjxxjlf0hu (60) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=as5a7a6c45 (55) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC (55) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 1 ACK (11) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1 (63) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Content-Length: 0 (17) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received ACK (6) - Command in SIP ACK [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #174 [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '3c26823294ed-t2wihiydodry@snom320-0004132496DC' of Response 1: Match Not Found <--- SIP read from 192.168.161.250:5060 ---> INVITE sip:701@pbx.example.net;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-4p532t0uc3d4;rport From: "Theo Belder" ;tag=bjxxjlf0hu To: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/6.5.10 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="403",realm="siprealm",nonce="1cdcdc03",uri="sip:701@pbx.example.net;user=phone",response="51d4e257bd8b955c9bb2ccb09f70e6b0",algorithm=MD5 Content-Type: application/sdp Content-Length: 271 v=0 o=root 490570670 490570670 IN IP4 192.168.161.250 s=call c=IN IP4 192.168.161.250 t=0 0 m=audio 15084 RTP/AVP 8 0 3 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:701@pbx.example.net;user=phone SIP/2.0 (49) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-4p532t0uc3d4;rport (71) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Theo Belder" ;tag=bjxxjlf0hu (60) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (40) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC (55) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 2 INVITE (14) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1 (63) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: P-Key-Flags: keys="3" (21) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: User-Agent: snom320/6.5.10 (26) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Accept: application/sdp (23) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: Session-Expires: 3600;refresher=uas (35) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 15: Min-SE: 90 (10) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 16: Proxy-Authorization: Digest username="403",realm="siprealm",nonce="1cdcdc03",uri="sip:701@pbx.example.net;user=phone",response="51d4e257bd8b955c9bb2ccb09f70e6b0",algorithm=MD5 (175) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 17: Content-Type: application/sdp (29) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 18: Content-Length: 271 (19) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 19: (0) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 490570670 490570670 IN IP4 192.168.161.250 (49) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.250 (24) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 15084 RTP/AVP 8 0 3 101 (31) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:0 pcmu/8000 (20) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:3 gsm/8000 (19) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (19 headers 13 lines) --- [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 192.168.161.250 : 5060 (NAT) Using INVITE request as basis request - 3c26823294ed-t2wihiydodry@snom320-0004132496DC [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:2624 do_setnat: Setting NAT on RTP to Off [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:2629 do_setnat: Setting NAT on VRTP to Off [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:2634 do_setnat: Setting NAT on UDPTL to Off Found user '403' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4993 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 192.168.161.250:15084 Found description format pcma for ID 8 Found description format pcmu for ID 0 Found description format gsm for ID 3 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x8 (alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.250:15084 [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:13549 handle_request_invite: Checking SIP call limits for device 403 [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:3058 update_call_counter: Updating call counter for incoming call [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:3133 update_call_counter: Call from peer '403' is 1 out of 30 [Sep 11 10:07:45] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/403 Looking for 701 in user-01 (domain pbx.example.net) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:3867 sip_new: *** Our native formats are 0x8 (alaw) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:3868 sip_new: *** Joint capabilities are 0x8 (alaw) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:3869 sip_new: *** Our capabilities are 0x8 (alaw) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:3870 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:3893 sip_new: This channel will not be able to handle video. [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:8065 build_route: build_route: Contact hop: ;flow-id=1 list_route: hop: [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:13624 handle_request_invite: SIP/403-09334c20: New call is still down.... Trying... <--- Transmitting (no NAT) to 192.168.161.250:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-4p532t0uc3d4;received=192.168.161.250;rport=5060 From: "Theo Belder" ;tag=bjxxjlf0hu To: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 2 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Sep 11 10:07:45] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:07:45] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/403-09334c20 [Sep 11 10:07:45] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:07:45] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/403 - state 2 (In use) [Sep 11 10:07:45] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:07:45] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:07:45] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:07:45] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:07:45] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/403 - state 2 (In use) [Sep 11 10:07:45] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:07:45] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:07:45] DEBUG[18501]: pbx.c:1809 pbx_extension_helper: Launching 'Dial'  -- Executing [701@user-01:1] Dial("SIP/403-09334c20", "SIP/401") in new stack [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:15556 sip_request_call: Asked to create a SIP channel with formats: 0x8 (alaw) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:2624 do_setnat: Setting NAT on RTP to Off [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:3867 sip_new: *** Our native formats are 0x8 (alaw) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:3868 sip_new: *** Joint capabilities are 0x0 (nothing) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:3869 sip_new: *** Our capabilities are 0x8 (alaw) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:3870 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:3872 sip_new: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:3893 sip_new: This channel will not be able to handle video. [Sep 11 10:07:45] DEBUG[18501]: rtp.c:1621 ast_rtp_make_compatible: Seeded SDP of 'SIP/401-0933a850' with that of 'SIP/403-09334c20' [Sep 11 10:07:45] DEBUG[18501]: channel.c:3514 ast_channel_inherit_variables: Not copying variable STACK-user-01-701-1. [Sep 11 10:07:45] DEBUG[18501]: channel.c:3514 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Sep 11 10:07:45] DEBUG[18501]: channel.c:3514 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Sep 11 10:07:45] DEBUG[18501]: channel.c:3514 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Sep 11 10:07:45] DEBUG[18501]: channel.c:3514 ast_channel_inherit_variables: Not copying variable SIPURI. [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:2882 sip_call: Outgoing Call for 401 [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:3058 update_call_counter: Updating call counter for outgoing call [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:3133 update_call_counter: Call to peer '401' is 1 out of 30 [Sep 11 10:07:45] DEBUG[18501]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/401 [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:2897 sip_call: Our T38 capability (3856), joint T38 capability (3856) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: False [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 192.168.161.100 port 15180 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:401@192.168.161.246:5060;line=4f6xat3o SIP/2.0 (57) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK63f8794a;rport (66) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 2: From: "Theo Belder" ;tag=as1b4cb1fd (60) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 3: To: (48) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 4: Contact: (34) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net (57) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 INVITE (16) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 9: Remote-Party-ID: "Theo Belder" ;privacy=off;screen=no (78) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 10: Date: Tue, 11 Sep 2007 08:07:45 GMT (35) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 12: Supported: replaces (19) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 13: Content-Type: application/sdp (29) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 14: Content-Length: 246 (19) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 15: (0) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: o=root 18440 18440 IN IP4 192.168.161.100 (41) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: s=session (9) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.100 (24) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: m=audio 15180 RTP/AVP 8 101 (27) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=silenceSupp:off - - - - (25) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.161.246:5060: INVITE sip:401@192.168.161.246:5060;line=4f6xat3o SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK63f8794a;rport From: "Theo Belder" ;tag=as1b4cb1fd To: Contact: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 102 INVITE User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Theo Belder" ;privacy=off;screen=no Date: Tue, 11 Sep 2007 08:07:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18440 IN IP4 192.168.161.100 s=session c=IN IP4 192.168.161.100 t=0 0 m=audio 15180 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 11 10:07:45] DEBUG[18501]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #176  -- Called 401 [Sep 11 10:07:45] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:45] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:45] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/401 - state 6 (Ringing) [Sep 11 10:07:45] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:45] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:45] DEBUG[18502]: app_queue.c:548 changethread: Device 'SIP/403' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Sep 11 10:07:45] DEBUG[18504]: app_queue.c:548 changethread: Device 'SIP/401' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Sep 11 10:07:45] DEBUG[18503]: app_queue.c:548 changethread: Device 'SIP/403' changed to state '2' (In use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.246:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK63f8794a;rport=5060 From: "Theo Belder" ;tag=as1b4cb1fd To: ;tag=ull0udxr9r Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK63f8794a;rport=5060 (71) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Theo Belder" ;tag=as1b4cb1fd (60) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=ull0udxr9r (63) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net (57) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 INVITE (16) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Content-Length: 0 (17) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:2171 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #176 - INVITE (got response) [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:2180 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0e3f4f4e700219793dcc419c77a3b373@pbx.example.net' Request 102: Found [Sep 11 10:07:45] DEBUG[18453]: chan_sip.c:11739 handle_response_invite: SIP response 180 to standard invite [Sep 11 10:07:45] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/401-0933a850  -- SIP/401-0933a850 is ringing [Sep 11 10:07:45] DEBUG[18501]: rtp.c:1550 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/403-09334c20' with that of 'SIP/401-0933a850' <--- Transmitting (no NAT) to 192.168.161.250:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-4p532t0uc3d4;received=192.168.161.250;rport=5060 From: "Theo Belder" ;tag=bjxxjlf0hu To: ;tag=as449bf3fa Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 2 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Sep 11 10:07:45] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:45] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:45] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/401 - state 6 (Ringing) [Sep 11 10:07:45] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:45] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:45] DEBUG[18505]: app_queue.c:548 changethread: Device 'SIP/401' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.246:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK63f8794a;rport=5060 From: "Theo Belder" ;tag=as1b4cb1fd To: ;tag=ull0udxr9r Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> [Sep 11 10:07:46] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Sep 11 10:07:46] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK63f8794a;rport=5060 (71) [Sep 11 10:07:46] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Theo Belder" ;tag=as1b4cb1fd (60) [Sep 11 10:07:46] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=ull0udxr9r (63) [Sep 11 10:07:46] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net (57) [Sep 11 10:07:46] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 INVITE (16) [Sep 11 10:07:46] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:07:46] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:46] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:46] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Content-Length: 0 (17) [Sep 11 10:07:46] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Sep 11 10:07:46] DEBUG[18453]: chan_sip.c:2180 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0e3f4f4e700219793dcc419c77a3b373@pbx.example.net' Request 102: Found [Sep 11 10:07:46] DEBUG[18453]: chan_sip.c:11739 handle_response_invite: SIP response 180 to standard invite  -- SIP/401-0933a850 is ringing [Sep 11 10:07:46] DEBUG[18501]: rtp.c:1550 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/403-09334c20' with that of 'SIP/401-0933a850' <--- SIP read from 192.168.161.246:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK63f8794a;rport=5060 From: "Theo Belder" ;tag=as1b4cb1fd To: ;tag=ull0udxr9r Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> [Sep 11 10:07:47] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Sep 11 10:07:47] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK63f8794a;rport=5060 (71) [Sep 11 10:07:47] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Theo Belder" ;tag=as1b4cb1fd (60) [Sep 11 10:07:47] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=ull0udxr9r (63) [Sep 11 10:07:47] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net (57) [Sep 11 10:07:47] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 INVITE (16) [Sep 11 10:07:47] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:07:47] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:47] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:47] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Content-Length: 0 (17) [Sep 11 10:07:47] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Sep 11 10:07:47] DEBUG[18453]: chan_sip.c:2180 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0e3f4f4e700219793dcc419c77a3b373@pbx.example.net' Request 102: Found [Sep 11 10:07:47] DEBUG[18453]: chan_sip.c:11739 handle_response_invite: SIP response 180 to standard invite  -- SIP/401-0933a850 is ringing [Sep 11 10:07:47] DEBUG[18501]: rtp.c:1550 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/403-09334c20' with that of 'SIP/401-0933a850' <--- SIP read from 192.168.161.246:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK63f8794a;rport=5060 From: "Theo Belder" ;tag=as1b4cb1fd To: ;tag=ull0udxr9r Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom360/6.5.10 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 224 v=0 o=root 295345829 295345830 IN IP4 192.168.161.246 s=call c=IN IP4 192.168.161.246 t=0 0 m=audio 19242 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 Ok (14) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK63f8794a;rport=5060 (71) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Theo Belder" ;tag=as1b4cb1fd (60) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=ull0udxr9r (63) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net (57) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 INVITE (16) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom360/6.5.10 (26) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Type: application/sdp (29) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 224 (19) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 295345829 295345830 IN IP4 192.168.161.246 (49) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.246 (24) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 19242 RTP/AVP 8 101 (27) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (13 headers 11 lines) --- [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:2120 __sip_ack: Acked pending invite 102 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '0e3f4f4e700219793dcc419c77a3b373@pbx.example.net' of Request 102: Match Not Found [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:11739 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.161.246:19242 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel SIP/401-0933a850 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.246:19242 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:5313 process_sdp: We have an owner, now see if we need to change this call [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:3058 update_call_counter: Updating call counter for outgoing call [Sep 11 10:07:48] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/401 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:8065 build_route: build_route: Contact hop: ;flow-id=1 list_route: hop: [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.246, port 5060 Transmitting (no NAT) to 192.168.161.246:5060: ACK sip:401@192.168.161.246:5060;line=4f6xat3o SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK5ca89563;rport From: "Theo Belder" ;tag=as1b4cb1fd To: ;tag=ull0udxr9r Contact: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 102 ACK User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Theo Belder" ;privacy=off;screen=no Content-Length: 0 --- [Sep 11 10:07:48] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:48] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:48] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/401 - state 2 (In use) [Sep 11 10:07:48] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:48] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:48] DEBUG[18501]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/401-0933a850  -- SIP/401-0933a850 answered SIP/403-09334c20 [Sep 11 10:07:48] DEBUG[18501]: rtp.c:1550 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/403-09334c20' with that of 'SIP/401-0933a850' [Sep 11 10:07:48] DEBUG[18501]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/403-09334c20 [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:3525 sip_answer: SIP answering channel: SIP/403-09334c20 [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:6499 transmit_response_with_sdp: Setting framing from config on incoming call [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.161.100 port 17674 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) <--- Reliably Transmitting (no NAT) to 192.168.161.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-4p532t0uc3d4;received=192.168.161.250;rport=5060 From: "Theo Belder" ;tag=bjxxjlf0hu To: ;tag=as449bf3fa Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 2 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18440 IN IP4 192.168.161.100 s=session c=IN IP4 192.168.161.100 t=0 0 m=audio 17674 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #178  -- Native bridging SIP/403-09334c20 and SIP/401-0933a850 [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:17158 sip_set_rtp_peer: Deferring reinvite on SIP '3c26823294ed-t2wihiydodry@snom320-0004132496DC' - It's audio will be redirected to IP 192.168.161.246 [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:17153 sip_set_rtp_peer: Sending reinvite on SIP '0e3f4f4e700219793dcc419c77a3b373@pbx.example.net' - It's audio soon redirected to IP 192.168.161.250 [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:5713 reqprep: Strict routing enforced for session 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.246, port 5060 [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 192.168.161.100 port 15180 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:1629 initialize_initreq: Initializing already initialized SIP dialog 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net (presumably reinvite) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:401@192.168.161.246:5060;line=4f6xat3o SIP/2.0 (57) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2e8a6b54;rport (66) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 2: From: "Theo Belder" ;tag=as1b4cb1fd (60) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=ull0udxr9r (63) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 4: Contact: (34) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net (57) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 6: CSeq: 103 INVITE (16) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 9: Remote-Party-ID: "Theo Belder" ;privacy=off;screen=no (78) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 12: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 13: Content-Type: application/sdp (29) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 14: Content-Length: 246 (19) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 15: (0) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: o=root 18440 18441 IN IP4 192.168.161.250 (41) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: s=session (9) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.250 (24) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: m=audio 15084 RTP/AVP 8 101 (27) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=silenceSupp:off - - - - (25) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.161.246:5060: INVITE sip:401@192.168.161.246:5060;line=4f6xat3o SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2e8a6b54;rport From: "Theo Belder" ;tag=as1b4cb1fd To: ;tag=ull0udxr9r Contact: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 103 INVITE User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Theo Belder" ;privacy=off;screen=no Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18441 IN IP4 192.168.161.250 s=session c=IN IP4 192.168.161.250 t=0 0 m=audio 15084 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 11 10:07:48] DEBUG[18501]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #179 [Sep 11 10:07:48] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:48] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:48] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/401 - state 2 (In use) [Sep 11 10:07:48] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:48] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:48] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:07:48] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:07:48] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/403 - state 2 (In use) [Sep 11 10:07:48] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:07:48] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:07:48] DEBUG[18506]: app_queue.c:548 changethread: Device 'SIP/401' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Sep 11 10:07:48] DEBUG[18507]: app_queue.c:548 changethread: Device 'SIP/401' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Sep 11 10:07:48] DEBUG[18508]: app_queue.c:548 changethread: Device 'SIP/403' changed to state '2' (In use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.250:5060 ---> ACK sip:701@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-l5a5ecmrovyg;rport From: "Theo Belder" ;tag=bjxxjlf0hu To: ;tag=as449bf3fa Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: ACK sip:701@192.168.161.100 SIP/2.0 (35) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-l5a5ecmrovyg;rport (71) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Theo Belder" ;tag=bjxxjlf0hu (60) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=as449bf3fa (55) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC (55) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 2 ACK (11) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1 (63) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Content-Length: 0 (17) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received ACK (6) - Command in SIP ACK [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #178 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '3c26823294ed-t2wihiydodry@snom320-0004132496DC' of Response 2: Match Not Found [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:11719 check_pendings: Sending pending reinvite on '3c26823294ed-t2wihiydodry@snom320-0004132496DC' [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 3c26823294ed-t2wihiydodry@snom320-0004132496DC set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.250, port 5060 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.161.100 port 17674 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:1629 initialize_initreq: Initializing already initialized SIP dialog 3c26823294ed-t2wihiydodry@snom320-0004132496DC (presumably reinvite) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:403@192.168.161.250:5060;line=xjqx46za SIP/2.0 (57) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK53107683;rport (66) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: ;tag=as449bf3fa (57) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Theo Belder" ;tag=bjxxjlf0hu (58) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (34) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC (55) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 INVITE (16) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Supported: replaces (19) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Type: application/sdp (29) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Content-Length: 246 (19) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: (0) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 18440 18441 IN IP4 192.168.161.246 (41) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=session (9) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.246 (24) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 19242 RTP/AVP 8 101 (27) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=silenceSupp:off - - - - (25) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.161.250:5060: INVITE sip:403@192.168.161.250:5060;line=xjqx46za SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK53107683;rport From: ;tag=as449bf3fa To: "Theo Belder" ;tag=bjxxjlf0hu Contact: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 102 INVITE User-Agent: atCOM PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18441 IN IP4 192.168.161.246 s=session c=IN IP4 192.168.161.246 t=0 0 m=audio 19242 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #180 [Sep 11 10:07:48] DEBUG[18501]: rtp.c:2727 ast_rtp_write: Ooh, format changed from unknown to alaw [Sep 11 10:07:48] DEBUG[18501]: rtp.c:2744 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160 <--- SIP read from 192.168.161.246:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2e8a6b54;rport=5060 From: "Theo Belder" ;tag=as1b4cb1fd To: ;tag=ull0udxr9r Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 103 INVITE Contact: ;flow-id=1 User-Agent: snom360/6.5.10 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 224 v=0 o=root 295345829 295345831 IN IP4 192.168.161.246 s=call c=IN IP4 192.168.161.246 t=0 0 m=audio 19242 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 Ok (14) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2e8a6b54;rport=5060 (71) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Theo Belder" ;tag=as1b4cb1fd (60) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=ull0udxr9r (63) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net (57) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 103 INVITE (16) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom360/6.5.10 (26) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Type: application/sdp (29) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 224 (19) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 295345829 295345831 IN IP4 192.168.161.246 (49) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.246 (24) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 19242 RTP/AVP 8 101 (27) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (13 headers 11 lines) --- [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:2120 __sip_ack: Acked pending invite 103 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #179 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '0e3f4f4e700219793dcc419c77a3b373@pbx.example.net' of Request 103: Match Not Found [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:11737 handle_response_invite: SIP response 200 to RE-invite on outgoing call 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.161.246:19242 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel SIP/401-0933a850 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.246:19242 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:5313 process_sdp: We have an owner, now see if we need to change this call [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:3058 update_call_counter: Updating call counter for outgoing call [Sep 11 10:07:48] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/401 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:8004 build_route: build_route: Retaining previous route: [Sep 11 10:07:48] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:48] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:48] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/401 - state 2 (In use) [Sep 11 10:07:48] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:48] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.246, port 5060 Transmitting (no NAT) to 192.168.161.246:5060: ACK sip:401@192.168.161.246:5060;line=4f6xat3o SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK35dd1b65;rport From: "Theo Belder" ;tag=as1b4cb1fd To: ;tag=ull0udxr9r Contact: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 103 ACK User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Theo Belder" ;privacy=off;screen=no Content-Length: 0 --- [Sep 11 10:07:48] DEBUG[18509]: app_queue.c:548 changethread: Device 'SIP/401' changed to state '2' (In use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.250:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK53107683;rport=5060 From: ;tag=as449bf3fa To: "Theo Belder" ;tag=bjxxjlf0hu Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom320/6.5.10 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 224 v=0 o=root 490570670 490570671 IN IP4 192.168.161.250 s=call c=IN IP4 192.168.161.250 t=0 0 m=audio 15084 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 Ok (14) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK53107683;rport=5060 (71) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: ;tag=as449bf3fa (57) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Theo Belder" ;tag=bjxxjlf0hu (58) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC (55) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 INVITE (16) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom320/6.5.10 (26) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Type: application/sdp (29) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 224 (19) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 490570670 490570671 IN IP4 192.168.161.250 (49) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.250 (24) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 15084 RTP/AVP 8 101 (27) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (13 headers 11 lines) --- [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:2120 __sip_ack: Acked pending invite 102 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #180 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '3c26823294ed-t2wihiydodry@snom320-0004132496DC' of Request 102: Match Not Found [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:11737 handle_response_invite: SIP response 200 to RE-invite on outgoing call 3c26823294ed-t2wihiydodry@snom320-0004132496DC Found RTP audio format 8 Found RTP audio format 101 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:4993 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 192.168.161.250:15084 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel SIP/403-09334c20 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.250:15084 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:5313 process_sdp: We have an owner, now see if we need to change this call [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:3058 update_call_counter: Updating call counter for incoming call [Sep 11 10:07:48] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/403 [Sep 11 10:07:48] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:07:48] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:07:48] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/403 - state 2 (In use) [Sep 11 10:07:48] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:07:48] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:8065 build_route: build_route: Contact hop: ;flow-id=1 list_route: hop: [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:11870 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:11875 handle_response_invite: T38 state changed to 0 on channel SIP [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:11878 handle_response_invite: T38 state changed to 0 on channel SIP/403-09334c20 [Sep 11 10:07:48] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 3c26823294ed-t2wihiydodry@snom320-0004132496DC set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.250, port 5060 Transmitting (no NAT) to 192.168.161.250:5060: ACK sip:403@192.168.161.250:5060;line=xjqx46za SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1278d82a;rport From: ;tag=as449bf3fa To: "Theo Belder" ;tag=bjxxjlf0hu Contact: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 102 ACK User-Agent: atCOM PBX Max-Forwards: 70 Content-Length: 0 --- [Sep 11 10:07:48] DEBUG[18510]: app_queue.c:548 changethread: Device 'SIP/403' changed to state '2' (In use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.2:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.2;branch=z9hG4bKac19305345 Max-Forwards: 70 From: ;tag=1c19296720 To: Call-ID: 9713438371120002023@192.168.161.2 CSeq: 73406 REGISTER Contact: ;expires=120 Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Authorization: Digest username="acM1000",realm="siprealm",nonce="760e1677",uri="sip:pbx.example.net",algorithm=MD5,response="6dc55638360dd05208dd71353792e6ed" Expires: 120 User-Agent: acM1000/v.4.80A.039 Content-Length: 0 <-------------> [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.2;branch=z9hG4bKac19305345 (55) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: Max-Forwards: 70 (16) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: From: ;tag=1c19296720 (61) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: To: (44) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 9713438371120002023@192.168.161.2 (42) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 73406 REGISTER (20) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;expires=120 (59) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Supported: em,timer,replaces,path (33) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Authorization: Digest username="acM1000",realm="siprealm",nonce="760e1677",uri="sip:pbx.example.net",algorithm=MD5,response="6dc55638360dd05208dd71353792e6ed" (158) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Expires: 120 (12) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: User-Agent: acM1000/v.4.80A.039 (31) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Content-Length: 0 (17) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: (0) --- (14 headers 0 lines) --- [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for 9713438371120002023@192.168.161.2 - REGISTER (No RTP) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.2 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.161.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.2;branch=z9hG4bKac19305345;received=192.168.161.2 From: ;tag=1c19296720 To: Call-ID: 9713438371120002023@192.168.161.2 CSeq: 73406 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.161.2:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.161.2;branch=z9hG4bKac19305345;received=192.168.161.2 From: ;tag=1c19296720 To: ;tag=as542e89b6 Call-ID: 9713438371120002023@192.168.161.2 CSeq: 73406 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="siprealm", nonce="720066a1" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '9713438371120002023@192.168.161.2' in 32000 ms (Method: REGISTER) <--- SIP read from 192.168.161.2:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.2;branch=z9hG4bKac19735827 Max-Forwards: 70 From: ;tag=1c19296720 To: Call-ID: 9713438371120002023@192.168.161.2 CSeq: 73407 REGISTER Contact: ;expires=120 Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Authorization: Digest username="acM1000",realm="siprealm",nonce="720066a1",uri="sip:pbx.example.net",algorithm=MD5,response="9ab017477c1b7d28ac03deb9d85962b4" Expires: 120 User-Agent: acM1000/v.4.80A.039 Content-Length: 0 <-------------> [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.2;branch=z9hG4bKac19735827 (55) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: Max-Forwards: 70 (16) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: From: ;tag=1c19296720 (61) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: To: (44) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 9713438371120002023@192.168.161.2 (42) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 73407 REGISTER (20) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;expires=120 (59) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Supported: em,timer,replaces,path (33) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Authorization: Digest username="acM1000",realm="siprealm",nonce="720066a1",uri="sip:pbx.example.net",algorithm=MD5,response="9ab017477c1b7d28ac03deb9d85962b4" (158) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Expires: 120 (12) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: User-Agent: acM1000/v.4.80A.039 (31) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Content-Length: 0 (17) [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: (0) --- (14 headers 0 lines) --- [Sep 11 10:07:49] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.2 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.161.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.2;branch=z9hG4bKac19735827;received=192.168.161.2 From: ;tag=1c19296720 To: Call-ID: 9713438371120002023@192.168.161.2 CSeq: 73407 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.161.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.2;branch=z9hG4bKac19735827;received=192.168.161.2 From: ;tag=1c19296720 To: ;tag=as542e89b6 Call-ID: 9713438371120002023@192.168.161.2 CSeq: 73407 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 120 Contact: ;expires=120 Date: Tue, 11 Sep 2007 08:07:49 GMT Content-Length: 0 <------------> [Sep 11 10:07:49] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/acM1000 Scheduling destruction of SIP dialog '9713438371120002023@192.168.161.2' in 32000 ms (Method: REGISTER) [Sep 11 10:07:49] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - acM1000 [Sep 11 10:07:49] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer acM1000 [Sep 11 10:07:49] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/acM1000 - state 1 (Not in use) [Sep 11 10:07:49] DEBUG[18511]: app_queue.c:548 changethread: Device 'SIP/acM1000' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.246:5060 ---> INVITE sip:403@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-3q7a7tmdy2of;rport From: ;tag=ull0udxr9r To: "Theo Belder" ;tag=as1b4cb1fd Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/6.5.10 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 271 v=0 o=root 295345829 295345832 IN IP4 192.168.161.246 s=call c=IN IP4 192.168.161.246 t=0 0 m=audio 19242 RTP/AVP 8 0 3 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendonly <-------------> [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:403@192.168.161.100 SIP/2.0 (38) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-3q7a7tmdy2of;rport (71) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: ;tag=ull0udxr9r (65) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Theo Belder" ;tag=as1b4cb1fd (58) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net (57) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 1 INVITE (14) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1 (63) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: P-Key-Flags: resolution="31x13", keys="4" (41) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: User-Agent: snom360/6.5.10 (26) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Accept: application/sdp (23) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: Session-Expires: 3600;refresher=uas (35) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 15: Min-SE: 90 (10) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 16: Content-Type: application/sdp (29) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 17: Content-Length: 271 (19) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 18: (0) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 295345829 295345832 IN IP4 192.168.161.246 (49) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.246 (24) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 19242 RTP/AVP 8 0 3 101 (31) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:0 pcmu/8000 (20) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:3 gsm/8000 (19) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendonly (10) --- (18 headers 13 lines) --- [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:1688 parse_sip_options: Begin: parsing SIP "Supported: timer, 100rel, replaces, callerid" [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:1696 parse_sip_options: Found SIP option: -timer- [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:1702 parse_sip_options: Matched SIP option: timer [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:1696 parse_sip_options: Found SIP option: -100rel- [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:1702 parse_sip_options: Matched SIP option: 100rel [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:1696 parse_sip_options: Found SIP option: -replaces- [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:1702 parse_sip_options: Matched SIP option: replaces [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:1696 parse_sip_options: Found SIP option: -callerid- [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:1710 parse_sip_options: Found no match for SIP option: callerid (Please file bug report!) Sending to 192.168.161.246 : 5060 (NAT) Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 192.168.161.246:19242 Found description format pcma for ID 8 Found description format pcmu for ID 0 Found description format gsm for ID 3 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel SIP/401-0933a850 Capabilities: us - 0x8 (alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.246:19242 [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:5313 process_sdp: We have an owner, now see if we need to change this call [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:13601 handle_request_invite: Got a SIP re-invite for call 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:13698 handle_request_invite: SIP/401-0933a850: This call is UP.... [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:6499 transmit_response_with_sdp: Setting framing from config on incoming call [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 192.168.161.100 port 15180 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:07:51] DEBUG[18453]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) <--- Transmitting (NAT) to 192.168.161.246:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-3q7a7tmdy2of;received=192.168.161.246;rport=5060 From: ;tag=ull0udxr9r To: "Theo Belder" ;tag=as1b4cb1fd Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 1 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18442 IN IP4 192.168.161.250 s=session c=IN IP4 192.168.161.250 t=0 0 m=audio 15084 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:17153 sip_set_rtp_peer: Sending reinvite on SIP '3c26823294ed-t2wihiydodry@snom320-0004132496DC' - It's audio soon redirected to IP 192.168.161.100 [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:5713 reqprep: Strict routing enforced for session 3c26823294ed-t2wihiydodry@snom320-0004132496DC set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.250, port 5060 [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.161.100 port 17674 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:1629 initialize_initreq: Initializing already initialized SIP dialog 3c26823294ed-t2wihiydodry@snom320-0004132496DC (presumably reinvite) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:403@192.168.161.250:5060;line=xjqx46za SIP/2.0 (57) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK516b650e;rport (66) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 2: From: ;tag=as449bf3fa (57) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 3: To: "Theo Belder" ;tag=bjxxjlf0hu (58) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 4: Contact: (34) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC (55) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 6: CSeq: 103 INVITE (16) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 10: Supported: replaces (19) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 12: Content-Type: application/sdp (29) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 13: Content-Length: 246 (19) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4640 parse_request: Header 14: (0) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: o=root 18440 18442 IN IP4 192.168.161.100 (41) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: s=session (9) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.100 (24) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: m=audio 17674 RTP/AVP 8 101 (27) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=silenceSupp:off - - - - (25) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.161.250:5060: INVITE sip:403@192.168.161.250:5060;line=xjqx46za SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK516b650e;rport From: ;tag=as449bf3fa To: "Theo Belder" ;tag=bjxxjlf0hu Contact: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 103 INVITE User-Agent: atCOM PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18442 IN IP4 192.168.161.100 s=session c=IN IP4 192.168.161.100 t=0 0 m=audio 17674 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #186  -- Started music on hold, class 'default', on SIP/403-09334c20 [Sep 11 10:07:51] DEBUG[18501]: rtp.c:2864 bridge_native_loop: Oooh, 'SIP/401-0933a850' changed end address to 0.0.0.0:0 (format 14) [Sep 11 10:07:51] DEBUG[18501]: rtp.c:2866 bridge_native_loop: Oooh, 'SIP/401-0933a850' changed end vaddress to 0.0.0.0:0 (format 14) [Sep 11 10:07:51] DEBUG[18501]: rtp.c:2868 bridge_native_loop: Oooh, 'SIP/401-0933a850' was 192.168.161.246:19242/(format 8) [Sep 11 10:07:51] DEBUG[18501]: rtp.c:2870 bridge_native_loop: Oooh, 'SIP/401-0933a850' was 0.0.0.0:0/(format 8) [Sep 11 10:07:51] DEBUG[18501]: chan_sip.c:17158 sip_set_rtp_peer: Deferring reinvite on SIP '3c26823294ed-t2wihiydodry@snom320-0004132496DC' - It's audio will be redirected to IP 192.168.161.100 <--- SIP read from 192.168.161.250:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK516b650e;rport=5060 From: ;tag=as449bf3fa To: "Theo Belder" ;tag=bjxxjlf0hu Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 103 INVITE Contact: ;flow-id=1 User-Agent: snom320/6.5.10 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 224 v=0 o=root 490570670 490570672 IN IP4 192.168.161.250 s=call c=IN IP4 192.168.161.250 t=0 0 m=audio 15084 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 Ok (14) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK516b650e;rport=5060 (71) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: ;tag=as449bf3fa (57) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Theo Belder" ;tag=bjxxjlf0hu (58) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC (55) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 103 INVITE (16) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom320/6.5.10 (26) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Type: application/sdp (29) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 224 (19) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 490570670 490570672 IN IP4 192.168.161.250 (49) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.250 (24) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 15084 RTP/AVP 8 101 (27) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (13 headers 11 lines) --- [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:2120 __sip_ack: Acked pending invite 103 [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #186 [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '3c26823294ed-t2wihiydodry@snom320-0004132496DC' of Request 103: Match Not Found [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:11737 handle_response_invite: SIP response 200 to RE-invite on outgoing call 3c26823294ed-t2wihiydodry@snom320-0004132496DC Found RTP audio format 8 Found RTP audio format 101 [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4993 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 192.168.161.250:15084 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel SIP/403-09334c20 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.250:15084 [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:5313 process_sdp: We have an owner, now see if we need to change this call [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:3058 update_call_counter: Updating call counter for incoming call [Sep 11 10:07:52] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/403 [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:8004 build_route: build_route: Retaining previous route: [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:11870 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:11875 handle_response_invite: T38 state changed to 0 on channel SIP [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:11878 handle_response_invite: T38 state changed to 0 on channel SIP/403-09334c20 [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 3c26823294ed-t2wihiydodry@snom320-0004132496DC set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.250, port 5060 Transmitting (no NAT) to 192.168.161.250:5060: ACK sip:403@192.168.161.250:5060;line=xjqx46za SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK66cbee65;rport From: ;tag=as449bf3fa To: "Theo Belder" ;tag=bjxxjlf0hu Contact: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 103 ACK User-Agent: atCOM PBX Max-Forwards: 70 Content-Length: 0 --- [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:11719 check_pendings: Sending pending reinvite on '3c26823294ed-t2wihiydodry@snom320-0004132496DC' [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 3c26823294ed-t2wihiydodry@snom320-0004132496DC set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.250, port 5060 [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.161.100 port 17674 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:1629 initialize_initreq: Initializing already initialized SIP dialog 3c26823294ed-t2wihiydodry@snom320-0004132496DC (presumably reinvite) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:403@192.168.161.250:5060;line=xjqx46za SIP/2.0 (57) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1036749e;rport (66) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: ;tag=as449bf3fa (57) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Theo Belder" ;tag=bjxxjlf0hu (58) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (34) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC (55) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 104 INVITE (16) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Supported: replaces (19) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Type: application/sdp (29) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Content-Length: 246 (19) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: (0) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 18440 18443 IN IP4 192.168.161.100 (41) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=session (9) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.100 (24) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 17674 RTP/AVP 8 101 (27) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=silenceSupp:off - - - - (25) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.161.250:5060: INVITE sip:403@192.168.161.250:5060;line=xjqx46za SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1036749e;rport From: ;tag=as449bf3fa To: "Theo Belder" ;tag=bjxxjlf0hu Contact: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 104 INVITE User-Agent: atCOM PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18443 IN IP4 192.168.161.100 s=session c=IN IP4 192.168.161.100 t=0 0 m=audio 17674 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #187 [Sep 11 10:07:52] DEBUG[18501]: res_musiconhold.c:261 ast_moh_files_next: SIP/403-09334c20 Opened file 1 '/var/lib/asterisk/moh/default/calm-river' [Sep 11 10:07:52] DEBUG[18501]: rtp.c:2597 ast_rtp_raw_write: Difference is 26600, ms is 3345 [Sep 11 10:07:52] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:07:52] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:07:52] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/403 - state 2 (In use) [Sep 11 10:07:52] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:07:52] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:07:52] DEBUG[18512]: app_queue.c:548 changethread: Device 'SIP/403' changed to state '2' (In use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.246:5060 ---> ACK sip:403@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-halobu0kbwmf;rport From: ;tag=ull0udxr9r To: "Theo Belder" ;tag=as1b4cb1fd Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: ACK sip:403@192.168.161.100 SIP/2.0 (35) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-halobu0kbwmf;rport (71) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: ;tag=ull0udxr9r (65) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Theo Belder" ;tag=as1b4cb1fd (58) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net (57) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 1 ACK (11) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1 (63) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Content-Length: 0 (17) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received ACK (6) - Command in SIP ACK [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '0e3f4f4e700219793dcc419c77a3b373@pbx.example.net' of Response 1: Match Found <--- SIP read from 192.168.161.250:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1036749e;rport=5060 From: ;tag=as449bf3fa To: "Theo Belder" ;tag=bjxxjlf0hu Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 104 INVITE Contact: ;flow-id=1 User-Agent: snom320/6.5.10 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 224 v=0 o=root 490570670 490570673 IN IP4 192.168.161.250 s=call c=IN IP4 192.168.161.250 t=0 0 m=audio 15084 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 Ok (14) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1036749e;rport=5060 (71) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: ;tag=as449bf3fa (57) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Theo Belder" ;tag=bjxxjlf0hu (58) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC (55) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 104 INVITE (16) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom320/6.5.10 (26) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Type: application/sdp (29) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 224 (19) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 490570670 490570673 IN IP4 192.168.161.250 (49) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.250 (24) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 15084 RTP/AVP 8 101 (27) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (13 headers 11 lines) --- [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:2120 __sip_ack: Acked pending invite 104 [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #187 [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '3c26823294ed-t2wihiydodry@snom320-0004132496DC' of Request 104: Match Not Found [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:11737 handle_response_invite: SIP response 200 to RE-invite on outgoing call 3c26823294ed-t2wihiydodry@snom320-0004132496DC Found RTP audio format 8 Found RTP audio format 101 [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:4993 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 192.168.161.250:15084 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel SIP/403-09334c20 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.250:15084 [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:5313 process_sdp: We have an owner, now see if we need to change this call [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:3058 update_call_counter: Updating call counter for incoming call [Sep 11 10:07:52] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/403 [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:8004 build_route: build_route: Retaining previous route: [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:11870 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:11875 handle_response_invite: T38 state changed to 0 on channel SIP [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:11878 handle_response_invite: T38 state changed to 0 on channel SIP/403-09334c20 [Sep 11 10:07:52] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 3c26823294ed-t2wihiydodry@snom320-0004132496DC set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.250, port 5060 Transmitting (no NAT) to 192.168.161.250:5060: ACK sip:403@192.168.161.250:5060;line=xjqx46za SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK217966d9;rport From: ;tag=as449bf3fa To: "Theo Belder" ;tag=bjxxjlf0hu Contact: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 104 ACK User-Agent: atCOM PBX Max-Forwards: 70 Content-Length: 0 --- [Sep 11 10:07:52] DEBUG[18501]: rtp.c:2597 ast_rtp_raw_write: Difference is 408, ms is 71 [Sep 11 10:07:52] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:07:52] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:07:52] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/403 - state 2 (In use) [Sep 11 10:07:52] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:07:52] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:07:52] DEBUG[18513]: app_queue.c:548 changethread: Device 'SIP/403' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Sep 11 10:07:53] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Sep 11 10:07:53] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: OPTIONS sip:192.168.1.160 SIP/2.0 (33) [Sep 11 10:07:53] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK514540ec;rport (66) [Sep 11 10:07:53] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as31857d6e (62) [Sep 11 10:07:53] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (23) [Sep 11 10:07:53] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (39) [Sep 11 10:07:53] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 7c2b0c2040c7001a5abc6a9576091c92@pbx.example.net (57) [Sep 11 10:07:53] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Sep 11 10:07:53] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:07:53] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:07:53] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Date: Tue, 11 Sep 2007 08:07:53 GMT (35) [Sep 11 10:07:53] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:07:53] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:07:53] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK514540ec;rport From: "asterisk" ;tag=as31857d6e To: Contact: Call-ID: 7c2b0c2040c7001a5abc6a9576091c92@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:07:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:07:53] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #188 [Sep 11 10:07:54] DEBUG[18453]: chan_sip.c:1879 retrans_pkt: SIP TIMER: Not rescheduling id #188:OPTIONS (Method 3) (No timer T1) Retransmitting #1 (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK514540ec;rport From: "asterisk" ;tag=as31857d6e To: Contact: Call-ID: 7c2b0c2040c7001a5abc6a9576091c92@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:07:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 192.168.161.246:5060 ---> INVITE sip:704@pbx.example.net;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-3fv9ctb1scvd;rport From: "Rámon" ;tag=pcfg2ibhgy To: Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/6.5.10 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 273 v=0 o=root 1313859764 1313859764 IN IP4 192.168.161.246 s=call c=IN IP4 192.168.161.246 t=0 0 m=audio 14868 RTP/AVP 8 0 3 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:704@pbx.example.net;user=phone SIP/2.0 (49) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-3fv9ctb1scvd;rport (71) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=pcfg2ibhgy (55) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (40) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 (55) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 1 INVITE (14) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1 (63) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: P-Key-Flags: resolution="31x13", keys="4" (41) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: User-Agent: snom360/6.5.10 (26) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Accept: application/sdp (23) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: Session-Expires: 3600;refresher=uas (35) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 15: Min-SE: 90 (10) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 16: Content-Type: application/sdp (29) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 17: Content-Length: 273 (19) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 18: (0) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 1313859764 1313859764 IN IP4 192.168.161.246 (51) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.246 (24) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 14868 RTP/AVP 8 0 3 101 (31) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:0 pcmu/8000 (20) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:3 gsm/8000 (19) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (18 headers 13 lines) --- [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:2624 do_setnat: Setting NAT on RTP to Off [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:2629 do_setnat: Setting NAT on VRTP to Off [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:2634 do_setnat: Setting NAT on UDPTL to Off [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 - INVITE (With RTP) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:1688 parse_sip_options: Begin: parsing SIP "Supported: timer, 100rel, replaces, callerid" [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:1696 parse_sip_options: Found SIP option: -timer- [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:1702 parse_sip_options: Matched SIP option: timer [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:1696 parse_sip_options: Found SIP option: -100rel- [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:1702 parse_sip_options: Matched SIP option: 100rel [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:1696 parse_sip_options: Found SIP option: -replaces- [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:1702 parse_sip_options: Matched SIP option: replaces [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:1696 parse_sip_options: Found SIP option: -callerid- [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:1710 parse_sip_options: Found no match for SIP option: callerid (Please file bug report!) Sending to 192.168.161.246 : 5060 (NAT) Using INVITE request as basis request - 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:2624 do_setnat: Setting NAT on RTP to Off [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:2629 do_setnat: Setting NAT on VRTP to Off [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:2634 do_setnat: Setting NAT on UDPTL to Off <--- Reliably Transmitting (no NAT) to 192.168.161.246:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-3fv9ctb1scvd;received=192.168.161.246;rport=5060 From: "Rámon" ;tag=pcfg2ibhgy To: ;tag=as0501e8bb Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 1 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="siprealm", nonce="2fa18d67" Content-Length: 0 <------------> [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #190 Scheduling destruction of SIP dialog '3c2681f30271-phvhk1y47mv2@snom360-000413231BC4' in 32000 ms (Method: INVITE) Found user '401' <--- SIP read from 192.168.161.246:5060 ---> ACK sip:704@pbx.example.net;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-3fv9ctb1scvd;rport From: "Rámon" ;tag=pcfg2ibhgy To: ;tag=as0501e8bb Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: ACK sip:704@pbx.example.net;user=phone SIP/2.0 (46) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-3fv9ctb1scvd;rport (71) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=pcfg2ibhgy (55) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=as0501e8bb (55) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 (55) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 1 ACK (11) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1 (63) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Content-Length: 0 (17) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received ACK (6) - Command in SIP ACK [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #190 [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '3c2681f30271-phvhk1y47mv2@snom360-000413231BC4' of Response 1: Match Not Found <--- SIP read from 192.168.161.246:5060 ---> INVITE sip:704@pbx.example.net;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-1q71b3hx2z1q;rport From: "Rámon" ;tag=pcfg2ibhgy To: Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/6.5.10 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="401",realm="siprealm",nonce="2fa18d67",uri="sip:704@pbx.example.net;user=phone",response="b59d84295aa9d2526925c85992c6be27",algorithm=MD5 Content-Type: application/sdp Content-Length: 273 v=0 o=root 1313859764 1313859764 IN IP4 192.168.161.246 s=call c=IN IP4 192.168.161.246 t=0 0 m=audio 14868 RTP/AVP 8 0 3 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:704@pbx.example.net;user=phone SIP/2.0 (49) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-1q71b3hx2z1q;rport (71) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=pcfg2ibhgy (55) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (40) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 (55) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 2 INVITE (14) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1 (63) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: P-Key-Flags: resolution="31x13", keys="4" (41) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: User-Agent: snom360/6.5.10 (26) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Accept: application/sdp (23) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: Session-Expires: 3600;refresher=uas (35) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 15: Min-SE: 90 (10) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 16: Proxy-Authorization: Digest username="401",realm="siprealm",nonce="2fa18d67",uri="sip:704@pbx.example.net;user=phone",response="b59d84295aa9d2526925c85992c6be27",algorithm=MD5 (175) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 17: Content-Type: application/sdp (29) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 18: Content-Length: 273 (19) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 19: (0) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 1313859764 1313859764 IN IP4 192.168.161.246 (51) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.246 (24) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 14868 RTP/AVP 8 0 3 101 (31) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:0 pcmu/8000 (20) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:3 gsm/8000 (19) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (19 headers 13 lines) --- [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 192.168.161.246 : 5060 (NAT) Using INVITE request as basis request - 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:2624 do_setnat: Setting NAT on RTP to Off [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:2629 do_setnat: Setting NAT on VRTP to Off [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:2634 do_setnat: Setting NAT on UDPTL to Off Found user '401' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4993 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 192.168.161.246:14868 Found description format pcma for ID 8 Found description format pcmu for ID 0 Found description format gsm for ID 3 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x8 (alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.246:14868 [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:13549 handle_request_invite: Checking SIP call limits for device 401 [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:3058 update_call_counter: Updating call counter for incoming call [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:3133 update_call_counter: Call from peer '401' is 2 out of 30 [Sep 11 10:07:55] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/401 Looking for 704 in user-01 (domain pbx.example.net) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:3867 sip_new: *** Our native formats are 0x8 (alaw) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:3868 sip_new: *** Joint capabilities are 0x8 (alaw) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:3869 sip_new: *** Our capabilities are 0x8 (alaw) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:3870 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:3893 sip_new: This channel will not be able to handle video. [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:8065 build_route: build_route: Contact hop: ;flow-id=1 list_route: hop: [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:13624 handle_request_invite: SIP/401-09355708: New call is still down.... Trying... <--- Transmitting (no NAT) to 192.168.161.246:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-1q71b3hx2z1q;received=192.168.161.246;rport=5060 From: "Rámon" ;tag=pcfg2ibhgy To: Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 2 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Sep 11 10:07:55] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/401-09355708 [Sep 11 10:07:55] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:55] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:55] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/401 - state 2 (In use) [Sep 11 10:07:55] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:55] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:55] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:55] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:55] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/401 - state 2 (In use) [Sep 11 10:07:55] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:55] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:55] DEBUG[18518]: pbx.c:1809 pbx_extension_helper: Launching 'Dial'  -- Executing [704@user-01:1] Dial("SIP/401-09355708", "SIP/404") in new stack [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:15556 sip_request_call: Asked to create a SIP channel with formats: 0x8 (alaw) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:2624 do_setnat: Setting NAT on RTP to Off [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:3867 sip_new: *** Our native formats are 0x8 (alaw) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:3868 sip_new: *** Joint capabilities are 0x0 (nothing) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:3869 sip_new: *** Our capabilities are 0x8 (alaw) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:3870 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:3872 sip_new: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:3893 sip_new: This channel will not be able to handle video. [Sep 11 10:07:55] DEBUG[18518]: rtp.c:1621 ast_rtp_make_compatible: Seeded SDP of 'SIP/404-0935a2b8' with that of 'SIP/401-09355708' [Sep 11 10:07:55] DEBUG[18518]: channel.c:3514 ast_channel_inherit_variables: Not copying variable STACK-user-01-704-1. [Sep 11 10:07:55] DEBUG[18518]: channel.c:3514 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Sep 11 10:07:55] DEBUG[18518]: channel.c:3514 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Sep 11 10:07:55] DEBUG[18518]: channel.c:3514 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Sep 11 10:07:55] DEBUG[18518]: channel.c:3514 ast_channel_inherit_variables: Not copying variable SIPURI. [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:2882 sip_call: Outgoing Call for 404 [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:3058 update_call_counter: Updating call counter for outgoing call [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:3133 update_call_counter: Call to peer '404' is 1 out of 30 [Sep 11 10:07:55] DEBUG[18518]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/404 [Sep 11 10:07:55] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:07:55] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:07:55] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/404 - state 6 (Ringing) [Sep 11 10:07:55] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:07:55] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:2897 sip_call: Our T38 capability (3856), joint T38 capability (3856) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: False [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 192.168.161.100 port 19310 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 (57) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK142ae571;rport (66) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=as456f0c98 (55) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 3: To: (48) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 4: Contact: (34) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net (57) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 INVITE (16) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 9: Remote-Party-ID: "Rámon" ;privacy=off;screen=no (73) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 10: Date: Tue, 11 Sep 2007 08:07:55 GMT (35) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 12: Supported: replaces (19) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 13: Content-Type: application/sdp (29) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 14: Content-Length: 246 (19) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 15: (0) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: o=root 18440 18440 IN IP4 192.168.161.100 (41) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: s=session (9) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.100 (24) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: m=audio 19310 RTP/AVP 8 101 (27) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=silenceSupp:off - - - - (25) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.161.247:5060: INVITE sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK142ae571;rport From: "Rámon" ;tag=as456f0c98 To: Contact: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 102 INVITE User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Rámon" ;privacy=off;screen=no Date: Tue, 11 Sep 2007 08:07:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18440 IN IP4 192.168.161.100 s=session c=IN IP4 192.168.161.100 t=0 0 m=audio 19310 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 11 10:07:55] DEBUG[18518]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #192  -- Called 404 [Sep 11 10:07:55] DEBUG[18519]: app_queue.c:548 changethread: Device 'SIP/401' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Sep 11 10:07:55] DEBUG[18520]: app_queue.c:548 changethread: Device 'SIP/401' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Sep 11 10:07:55] DEBUG[18521]: app_queue.c:548 changethread: Device 'SIP/404' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.247:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK142ae571;rport=5060 From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK142ae571;rport=5060 (71) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=as456f0c98 (55) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=viejglm0sj (63) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net (57) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 INVITE (16) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Content-Length: 0 (17) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:2171 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #192 - INVITE (got response) [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:2180 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '16fb24774565e40c06ce92b510a92129@pbx.example.net' Request 102: Found [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:11739 handle_response_invite: SIP response 180 to standard invite [Sep 11 10:07:55] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/404-0935a2b8 [Sep 11 10:07:55] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:07:55] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:07:55] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/404 - state 6 (Ringing) [Sep 11 10:07:55] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:07:55] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404  -- SIP/404-0935a2b8 is ringing [Sep 11 10:07:55] DEBUG[18522]: app_queue.c:548 changethread: Device 'SIP/404' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Sep 11 10:07:55] DEBUG[18518]: rtp.c:1550 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/401-09355708' with that of 'SIP/404-0935a2b8' <--- Transmitting (no NAT) to 192.168.161.246:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-1q71b3hx2z1q;received=192.168.161.246;rport=5060 From: "Rámon" ;tag=pcfg2ibhgy To: ;tag=as5d5ffe6e Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 2 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Sep 11 10:07:55] DEBUG[18453]: chan_sip.c:1879 retrans_pkt: SIP TIMER: Not rescheduling id #188:OPTIONS (Method 3) (No timer T1) Retransmitting #2 (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK514540ec;rport From: "asterisk" ;tag=as31857d6e To: Contact: Call-ID: 7c2b0c2040c7001a5abc6a9576091c92@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:07:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 192.168.161.247:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK142ae571;rport=5060 From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> [Sep 11 10:07:56] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Sep 11 10:07:56] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK142ae571;rport=5060 (71) [Sep 11 10:07:56] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=as456f0c98 (55) [Sep 11 10:07:56] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=viejglm0sj (63) [Sep 11 10:07:56] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net (57) [Sep 11 10:07:56] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 INVITE (16) [Sep 11 10:07:56] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:07:56] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:56] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:56] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Content-Length: 0 (17) [Sep 11 10:07:56] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Sep 11 10:07:56] DEBUG[18453]: chan_sip.c:2180 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '16fb24774565e40c06ce92b510a92129@pbx.example.net' Request 102: Found [Sep 11 10:07:56] DEBUG[18453]: chan_sip.c:11739 handle_response_invite: SIP response 180 to standard invite  -- SIP/404-0935a2b8 is ringing [Sep 11 10:07:56] DEBUG[18518]: rtp.c:1550 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/401-09355708' with that of 'SIP/404-0935a2b8' [Sep 11 10:07:56] DEBUG[18453]: chan_sip.c:1879 retrans_pkt: SIP TIMER: Not rescheduling id #188:OPTIONS (Method 3) (No timer T1) Retransmitting #3 (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK514540ec;rport From: "asterisk" ;tag=as31857d6e To: Contact: Call-ID: 7c2b0c2040c7001a5abc6a9576091c92@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:07:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 192.168.161.247:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK142ae571;rport=5060 From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> [Sep 11 10:07:57] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Sep 11 10:07:57] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK142ae571;rport=5060 (71) [Sep 11 10:07:57] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=as456f0c98 (55) [Sep 11 10:07:57] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=viejglm0sj (63) [Sep 11 10:07:57] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net (57) [Sep 11 10:07:57] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 INVITE (16) [Sep 11 10:07:57] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:07:57] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:57] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:57] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Content-Length: 0 (17) [Sep 11 10:07:57] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Sep 11 10:07:57] DEBUG[18453]: chan_sip.c:2180 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '16fb24774565e40c06ce92b510a92129@pbx.example.net' Request 102: Found [Sep 11 10:07:57] DEBUG[18453]: chan_sip.c:11739 handle_response_invite: SIP response 180 to standard invite  -- SIP/404-0935a2b8 is ringing [Sep 11 10:07:57] DEBUG[18518]: rtp.c:1550 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/401-09355708' with that of 'SIP/404-0935a2b8' [Sep 11 10:07:57] DEBUG[18453]: chan_sip.c:1879 retrans_pkt: SIP TIMER: Not rescheduling id #188:OPTIONS (Method 3) (No timer T1) Retransmitting #4 (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK514540ec;rport From: "asterisk" ;tag=as31857d6e To: Contact: Call-ID: 7c2b0c2040c7001a5abc6a9576091c92@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:07:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:07:57] DEBUG[18453]: chan_sip.c:3165 sip_destroy: Destroying SIP dialog 7c2b0c2040c7001a5abc6a9576091c92@pbx.example.net Really destroying SIP dialog '7c2b0c2040c7001a5abc6a9576091c92@pbx.example.net' Method: OPTIONS [Sep 11 10:07:57] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/EiconSoftIP [Sep 11 10:07:57] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - EiconSoftIP [Sep 11 10:07:57] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer EiconSoftIP [Sep 11 10:07:57] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/EiconSoftIP - state 1 (Not in use) [Sep 11 10:07:57] DEBUG[18523]: app_queue.c:548 changethread: Device 'SIP/EiconSoftIP' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.247:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK142ae571;rport=5060 From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK142ae571;rport=5060 (71) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=as456f0c98 (55) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=viejglm0sj (63) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net (57) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 INVITE (16) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Content-Length: 0 (17) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:2180 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '16fb24774565e40c06ce92b510a92129@pbx.example.net' Request 102: Found [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:11739 handle_response_invite: SIP response 180 to standard invite  -- SIP/404-0935a2b8 is ringing [Sep 11 10:07:59] DEBUG[18518]: rtp.c:1550 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/401-09355708' with that of 'SIP/404-0935a2b8' <--- SIP read from 192.168.161.247:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK142ae571;rport=5060 From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom360/6.5.10 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 224 v=0 o=root 500070299 500070300 IN IP4 192.168.161.247 s=call c=IN IP4 192.168.161.247 t=0 0 m=audio 18596 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 Ok (14) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK142ae571;rport=5060 (71) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=as456f0c98 (55) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=viejglm0sj (63) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net (57) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 INVITE (16) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom360/6.5.10 (26) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Type: application/sdp (29) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 224 (19) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 500070299 500070300 IN IP4 192.168.161.247 (49) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.247 (24) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 18596 RTP/AVP 8 101 (27) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (13 headers 11 lines) --- [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:2120 __sip_ack: Acked pending invite 102 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '16fb24774565e40c06ce92b510a92129@pbx.example.net' of Request 102: Match Not Found [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:11739 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.161.247:18596 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel SIP/404-0935a2b8 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.247:18596 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:5313 process_sdp: We have an owner, now see if we need to change this call [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:3058 update_call_counter: Updating call counter for outgoing call [Sep 11 10:07:59] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/404 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:8065 build_route: build_route: Contact hop: ;flow-id=1 list_route: hop: [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 16fb24774565e40c06ce92b510a92129@pbx.example.net set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.247, port 5060 Transmitting (no NAT) to 192.168.161.247:5060: ACK sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1d65c6a3;rport From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Contact: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 102 ACK User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Rámon" ;privacy=off;screen=no Content-Length: 0 --- [Sep 11 10:07:59] DEBUG[18518]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/404-0935a2b8  -- SIP/404-0935a2b8 answered SIP/401-09355708 [Sep 11 10:07:59] DEBUG[18518]: rtp.c:1550 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/401-09355708' with that of 'SIP/404-0935a2b8' [Sep 11 10:07:59] DEBUG[18518]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/401-09355708 [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:3525 sip_answer: SIP answering channel: SIP/401-09355708 [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:6499 transmit_response_with_sdp: Setting framing from config on incoming call [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.161.100 port 14096 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) <--- Reliably Transmitting (no NAT) to 192.168.161.246:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-1q71b3hx2z1q;received=192.168.161.246;rport=5060 From: "Rámon" ;tag=pcfg2ibhgy To: ;tag=as5d5ffe6e Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 2 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18440 IN IP4 192.168.161.100 s=session c=IN IP4 192.168.161.100 t=0 0 m=audio 14096 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #195  -- Native bridging SIP/401-09355708 and SIP/404-0935a2b8 [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:17158 sip_set_rtp_peer: Deferring reinvite on SIP '3c2681f30271-phvhk1y47mv2@snom360-000413231BC4' - It's audio will be redirected to IP 192.168.161.247 [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:17153 sip_set_rtp_peer: Sending reinvite on SIP '16fb24774565e40c06ce92b510a92129@pbx.example.net' - It's audio soon redirected to IP 192.168.161.246 [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:5713 reqprep: Strict routing enforced for session 16fb24774565e40c06ce92b510a92129@pbx.example.net set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.247, port 5060 [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 192.168.161.100 port 19310 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:1629 initialize_initreq: Initializing already initialized SIP dialog 16fb24774565e40c06ce92b510a92129@pbx.example.net (presumably reinvite) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 (57) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK51072655;rport (66) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=as456f0c98 (55) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=viejglm0sj (63) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 4: Contact: (34) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net (57) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 6: CSeq: 103 INVITE (16) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 9: Remote-Party-ID: "Rámon" ;privacy=off;screen=no (73) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 12: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 13: Content-Type: application/sdp (29) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 14: Content-Length: 246 (19) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 15: (0) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: o=root 18440 18441 IN IP4 192.168.161.246 (41) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: s=session (9) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.246 (24) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: m=audio 14868 RTP/AVP 8 101 (27) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=silenceSupp:off - - - - (25) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.161.247:5060: INVITE sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK51072655;rport From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Contact: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 103 INVITE User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Rámon" ;privacy=off;screen=no Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18441 IN IP4 192.168.161.246 s=session c=IN IP4 192.168.161.246 t=0 0 m=audio 14868 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #196 [Sep 11 10:07:59] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:07:59] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:07:59] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/404 - state 2 (In use) [Sep 11 10:07:59] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:07:59] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:07:59] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:07:59] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:07:59] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/404 - state 2 (In use) [Sep 11 10:07:59] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:07:59] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:07:59] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:59] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:59] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/401 - state 2 (In use) [Sep 11 10:07:59] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:59] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:59] DEBUG[18524]: app_queue.c:548 changethread: Device 'SIP/404' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Sep 11 10:07:59] DEBUG[18525]: app_queue.c:548 changethread: Device 'SIP/404' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Sep 11 10:07:59] DEBUG[18526]: app_queue.c:548 changethread: Device 'SIP/401' changed to state '2' (In use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.246:5060 ---> ACK sip:704@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-90ujrguexb1g;rport From: "Rámon" ;tag=pcfg2ibhgy To: ;tag=as5d5ffe6e Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: ACK sip:704@192.168.161.100 SIP/2.0 (35) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-90ujrguexb1g;rport (71) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=pcfg2ibhgy (55) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=as5d5ffe6e (55) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 (55) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 2 ACK (11) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1 (63) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Content-Length: 0 (17) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received ACK (6) - Command in SIP ACK [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #195 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '3c2681f30271-phvhk1y47mv2@snom360-000413231BC4' of Response 2: Match Not Found [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:11719 check_pendings: Sending pending reinvite on '3c2681f30271-phvhk1y47mv2@snom360-000413231BC4' [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.246, port 5060 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.161.100 port 14096 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:1629 initialize_initreq: Initializing already initialized SIP dialog 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 (presumably reinvite) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:401@192.168.161.246:5060;line=4f6xat3o SIP/2.0 (57) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK6d9654d2;rport (66) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: ;tag=as5d5ffe6e (57) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Rámon" ;tag=pcfg2ibhgy (53) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (34) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 (55) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 INVITE (16) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Supported: replaces (19) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Type: application/sdp (29) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Content-Length: 246 (19) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: (0) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 18440 18441 IN IP4 192.168.161.247 (41) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=session (9) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.247 (24) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 18596 RTP/AVP 8 101 (27) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=silenceSupp:off - - - - (25) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.161.246:5060: INVITE sip:401@192.168.161.246:5060;line=4f6xat3o SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK6d9654d2;rport From: ;tag=as5d5ffe6e To: "Rámon" ;tag=pcfg2ibhgy Contact: Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 102 INVITE User-Agent: atCOM PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18441 IN IP4 192.168.161.247 s=session c=IN IP4 192.168.161.247 t=0 0 m=audio 18596 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #197 [Sep 11 10:07:59] DEBUG[18518]: chan_sip.c:4151 sip_rtp_read: Bogus frame of format 'ulaw' received from 'SIP/404-0935a2b8'! [Sep 11 10:07:59] DEBUG[18518]: rtp.c:2727 ast_rtp_write: Ooh, format changed from unknown to alaw [Sep 11 10:07:59] DEBUG[18518]: rtp.c:2744 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160 [Sep 11 10:07:59] DEBUG[18518]: rtp.c:2727 ast_rtp_write: Ooh, format changed from unknown to alaw [Sep 11 10:07:59] DEBUG[18518]: rtp.c:2744 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160 <--- SIP read from 192.168.161.246:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK6d9654d2;rport=5060 From: ;tag=as5d5ffe6e To: "Rámon" ;tag=pcfg2ibhgy Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom360/6.5.10 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 226 v=0 o=root 1313859764 1313859765 IN IP4 192.168.161.246 s=call c=IN IP4 192.168.161.246 t=0 0 m=audio 14868 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 Ok (14) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK6d9654d2;rport=5060 (71) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: ;tag=as5d5ffe6e (57) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Rámon" ;tag=pcfg2ibhgy (53) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 (55) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 INVITE (16) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom360/6.5.10 (26) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Type: application/sdp (29) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 226 (19) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 1313859764 1313859765 IN IP4 192.168.161.246 (51) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.246 (24) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 14868 RTP/AVP 8 101 (27) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (13 headers 11 lines) --- [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:2120 __sip_ack: Acked pending invite 102 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #197 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '3c2681f30271-phvhk1y47mv2@snom360-000413231BC4' of Request 102: Match Not Found [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:11737 handle_response_invite: SIP response 200 to RE-invite on outgoing call 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 Found RTP audio format 8 Found RTP audio format 101 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4993 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 192.168.161.246:14868 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel SIP/401-09355708 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.246:14868 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:5313 process_sdp: We have an owner, now see if we need to change this call [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:3058 update_call_counter: Updating call counter for incoming call [Sep 11 10:07:59] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/401 [Sep 11 10:07:59] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:59] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:59] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/401 - state 2 (In use) [Sep 11 10:07:59] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:07:59] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:8065 build_route: build_route: Contact hop: ;flow-id=1 list_route: hop: [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:11870 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:11875 handle_response_invite: T38 state changed to 0 on channel SIP [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:11878 handle_response_invite: T38 state changed to 0 on channel SIP/401-09355708 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.246, port 5060 Transmitting (no NAT) to 192.168.161.246:5060: ACK sip:401@192.168.161.246:5060;line=4f6xat3o SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK42766df3;rport From: ;tag=as5d5ffe6e To: "Rámon" ;tag=pcfg2ibhgy Contact: Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 102 ACK User-Agent: atCOM PBX Max-Forwards: 70 Content-Length: 0 --- [Sep 11 10:07:59] DEBUG[18527]: app_queue.c:548 changethread: Device 'SIP/401' changed to state '2' (In use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.247:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK51072655;rport=5060 From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 103 INVITE Contact: ;flow-id=1 User-Agent: snom360/6.5.10 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 224 v=0 o=root 500070299 500070301 IN IP4 192.168.161.247 s=call c=IN IP4 192.168.161.247 t=0 0 m=audio 18596 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 Ok (14) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK51072655;rport=5060 (71) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=as456f0c98 (55) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=viejglm0sj (63) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net (57) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 103 INVITE (16) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom360/6.5.10 (26) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow-Events: talk, hold, refer (31) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Type: application/sdp (29) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 224 (19) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 500070299 500070301 IN IP4 192.168.161.247 (49) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.247 (24) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 18596 RTP/AVP 8 101 (27) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (13 headers 11 lines) --- [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:2120 __sip_ack: Acked pending invite 103 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #196 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '16fb24774565e40c06ce92b510a92129@pbx.example.net' of Request 103: Match Not Found [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:11737 handle_response_invite: SIP response 200 to RE-invite on outgoing call 16fb24774565e40c06ce92b510a92129@pbx.example.net Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.161.247:18596 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel SIP/404-0935a2b8 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.247:18596 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:5313 process_sdp: We have an owner, now see if we need to change this call [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:3058 update_call_counter: Updating call counter for outgoing call [Sep 11 10:07:59] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/404 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:8004 build_route: build_route: Retaining previous route: [Sep 11 10:07:59] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:07:59] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:07:59] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/404 - state 2 (In use) [Sep 11 10:07:59] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:07:59] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:07:59] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 16fb24774565e40c06ce92b510a92129@pbx.example.net set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.247, port 5060 Transmitting (no NAT) to 192.168.161.247:5060: ACK sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK3e6cfdc5;rport From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Contact: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 103 ACK User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Rámon" ;privacy=off;screen=no Content-Length: 0 --- [Sep 11 10:07:59] DEBUG[18528]: app_queue.c:548 changethread: Device 'SIP/404' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: OPTIONS sip:405@192.168.161.238:5060 SIP/2.0 (44) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK56d03d9f;rport (66) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as1228124f (62) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (34) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (39) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 0472287f4539289a1ff5932f15feafc1@pbx.example.net (57) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Date: Tue, 11 Sep 2007 08:08:00 GMT (35) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 192.168.161.238:5060: OPTIONS sip:405@192.168.161.238:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK56d03d9f;rport From: "asterisk" ;tag=as1228124f To: Contact: Call-ID: 0472287f4539289a1ff5932f15feafc1@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #200 [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: OPTIONS sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 (58) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK58c9339d;rport (66) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as2b5e02b6 (62) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (48) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (39) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 4f714b972e06b603284282d34ef39cfe@pbx.example.net (57) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Date: Tue, 11 Sep 2007 08:08:00 GMT (35) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 192.168.161.247:5060: OPTIONS sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK58c9339d;rport From: "asterisk" ;tag=as2b5e02b6 To: Contact: Call-ID: 4f714b972e06b603284282d34ef39cfe@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #202 <--- SIP read from 192.168.161.247:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK58c9339d;rport=5060 From: "asterisk" ;tag=as2b5e02b6 To: Call-ID: 4f714b972e06b603284282d34ef39cfe@pbx.example.net CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom360/6.5.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Length: 0 <-------------> [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 OK (14) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK58c9339d;rport=5060 (71) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as2b5e02b6 (62) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (48) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 4f714b972e06b603284282d34ef39cfe@pbx.example.net (57) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom360/6.5.10 (26) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Accept-Language: en (19) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Accept: application/sdp (23) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Allow-Events: talk, hold, refer (31) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Content-Length: 0 (17) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: (0) --- (14 headers 0 lines) --- [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #202 [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '4f714b972e06b603284282d34ef39cfe@pbx.example.net' of Request 102: Match Not Found Really destroying SIP dialog '4f714b972e06b603284282d34ef39cfe@pbx.example.net' Method: OPTIONS <--- SIP read from 192.168.161.238:5060 ---> SIP/2.0 200 OK From: "asterisk";tag=as1228124f To: ;tag=eea1a8c0-13c4-14826-50144f0-e64 Call-ID: 0472287f4539289a1ff5932f15feafc1@pbx.example.net CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.161.100:5060;rport=5060;branch=z9hG4bK56d03d9f Supported: replaces,100rel,timer Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS, INFO, PRACK, UPDATE User-Agent: Swissvoice IP10 SP v1.0.1 (Build 4) 3.0.5.1 Accept: application/sdp Content-Length: 0 <-------------> [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 OK (14) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: From: "asterisk";tag=as1228124f (61) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: To: ;tag=eea1a8c0-13c4-14826-50144f0-e64 (70) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: Call-ID: 0472287f4539289a1ff5932f15feafc1@pbx.example.net (57) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Via: SIP/2.0/UDP 192.168.161.100:5060;rport=5060;branch=z9hG4bK56d03d9f (71) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Supported: replaces,100rel,timer (32) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS, INFO, PRACK, UPDATE (76) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: User-Agent: Swissvoice IP10 SP v1.0.1 (Build 4) 3.0.5.1 (55) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Accept: application/sdp (23) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Content-Length: 0 (17) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: (0) --- (11 headers 0 lines) --- [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #200 [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '0472287f4539289a1ff5932f15feafc1@pbx.example.net' of Request 102: Match Not Found Really destroying SIP dialog '0472287f4539289a1ff5932f15feafc1@pbx.example.net' Method: OPTIONS [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: OPTIONS sip:403@192.168.161.250:5060;line=xjqx46za SIP/2.0 (58) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK3d409fa5;rport (66) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as6a58fd4a (62) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (48) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (39) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 4d277aec04f0e70208c685684bed0ff7@pbx.example.net (57) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Date: Tue, 11 Sep 2007 08:08:00 GMT (35) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 192.168.161.250:5060: OPTIONS sip:403@192.168.161.250:5060;line=xjqx46za SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK3d409fa5;rport From: "asterisk" ;tag=as6a58fd4a To: Contact: Call-ID: 4d277aec04f0e70208c685684bed0ff7@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:00] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #206 <--- SIP read from 192.168.161.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK3d409fa5;rport=5060 From: "asterisk" ;tag=as6a58fd4a To: Call-ID: 4d277aec04f0e70208c685684bed0ff7@pbx.example.net CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom320/6.5.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Length: 0 <-------------> [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 OK (14) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK3d409fa5;rport=5060 (71) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as6a58fd4a (62) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (48) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 4d277aec04f0e70208c685684bed0ff7@pbx.example.net (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom320/6.5.10 (26) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Accept-Language: en (19) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Accept: application/sdp (23) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Allow-Events: talk, hold, refer (31) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Content-Length: 0 (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: (0) --- (14 headers 0 lines) --- [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #206 [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '4d277aec04f0e70208c685684bed0ff7@pbx.example.net' of Request 102: Match Not Found Really destroying SIP dialog '4d277aec04f0e70208c685684bed0ff7@pbx.example.net' Method: OPTIONS [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: OPTIONS sip:401@192.168.161.246:5060;line=4f6xat3o SIP/2.0 (58) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK7b6b444f;rport (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as67171d0e (62) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (48) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (39) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 3eecb12032ca19b149c53e280f2b2235@pbx.example.net (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Date: Tue, 11 Sep 2007 08:08:01 GMT (35) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 192.168.161.246:5060: OPTIONS sip:401@192.168.161.246:5060;line=4f6xat3o SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK7b6b444f;rport From: "asterisk" ;tag=as67171d0e To: Contact: Call-ID: 3eecb12032ca19b149c53e280f2b2235@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #209 <--- SIP read from 192.168.161.246:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK7b6b444f;rport=5060 From: "asterisk" ;tag=as67171d0e To: Call-ID: 3eecb12032ca19b149c53e280f2b2235@pbx.example.net CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom360/6.5.10 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Length: 0 <-------------> [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 OK (14) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK7b6b444f;rport=5060 (71) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as67171d0e (62) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (48) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3eecb12032ca19b149c53e280f2b2235@pbx.example.net (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom360/6.5.10 (26) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Accept-Language: en (19) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Accept: application/sdp (23) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Allow-Events: talk, hold, refer (31) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Content-Length: 0 (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: (0) --- (14 headers 0 lines) --- [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #209 [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '3eecb12032ca19b149c53e280f2b2235@pbx.example.net' of Request 102: Match Not Found Really destroying SIP dialog '3eecb12032ca19b149c53e280f2b2235@pbx.example.net' Method: OPTIONS [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: OPTIONS sip:104@192.168.161.241;user=phone SIP/2.0 (50) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK59943b12;rport (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as0b7a80f4 (62) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (40) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (39) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 5a4065bb26a346b129226b615b1af972@pbx.example.net (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Date: Tue, 11 Sep 2007 08:08:01 GMT (35) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 192.168.161.241:5060: OPTIONS sip:104@192.168.161.241;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK59943b12;rport From: "asterisk" ;tag=as0b7a80f4 To: Contact: Call-ID: 5a4065bb26a346b129226b615b1af972@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #212 <--- SIP read from 192.168.161.241:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK59943b12;rport From: "asterisk" ;tag=as0b7a80f4 To: ;tag=1c199582093 Call-ID: 5a4065bb26a346b129226b615b1af972@pbx.example.net CSeq: 102 OPTIONS Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE X-Resources: telchs=4/0;mediachs=0/0 Content-Type: application/sdp Content-Length: 238 v=0 o=AudiocodesGW 199586304 199586179 IN IP4 192.168.161.241 s=Phone-Call c=IN IP4 192.168.161.241 t=0 0 m=audio 10000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 OK (14) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK59943b12;rport (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as0b7a80f4 (62) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=1c199582093 (56) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 5a4065bb26a346b129226b615b1af972@pbx.example.net (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Supported: em,100rel,timer,replaces,path,resource-priority (58) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: X-Resources: telchs=4/0;mediachs=0/0 (36) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Content-Type: application/sdp (29) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Content-Length: 238 (19) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: (0) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=AudiocodesGW 199586304 199586179 IN IP4 192.168.161.241 (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=Phone-Call (12) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.241 (24) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 10000 RTP/AVP 8 101 (27) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-15 (15) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (11 headers 11 lines) --- [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #212 [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '5a4065bb26a346b129226b615b1af972@pbx.example.net' of Request 102: Match Not Found Really destroying SIP dialog '5a4065bb26a346b129226b615b1af972@pbx.example.net' Method: OPTIONS [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: OPTIONS sip:103@192.168.161.241;user=phone SIP/2.0 (50) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2d97e460;rport (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as395b9de5 (62) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (40) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (39) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 6b2e2d1e5a9a08ae3b3c9d8f4d64afdf@pbx.example.net (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Date: Tue, 11 Sep 2007 08:08:01 GMT (35) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 192.168.161.241:5060: OPTIONS sip:103@192.168.161.241;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2d97e460;rport From: "asterisk" ;tag=as395b9de5 To: Contact: Call-ID: 6b2e2d1e5a9a08ae3b3c9d8f4d64afdf@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #215 <--- SIP read from 192.168.161.241:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2d97e460;rport From: "asterisk" ;tag=as395b9de5 To: ;tag=1c199833255 Call-ID: 6b2e2d1e5a9a08ae3b3c9d8f4d64afdf@pbx.example.net CSeq: 102 OPTIONS Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE X-Resources: telchs=4/0;mediachs=0/0 Content-Type: application/sdp Content-Length: 238 v=0 o=AudiocodesGW 199836380 199836257 IN IP4 192.168.161.241 s=Phone-Call c=IN IP4 192.168.161.241 t=0 0 m=audio 10000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 OK (14) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2d97e460;rport (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as395b9de5 (62) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=1c199833255 (56) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 6b2e2d1e5a9a08ae3b3c9d8f4d64afdf@pbx.example.net (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Supported: em,100rel,timer,replaces,path,resource-priority (58) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: X-Resources: telchs=4/0;mediachs=0/0 (36) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Content-Type: application/sdp (29) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Content-Length: 238 (19) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: (0) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=AudiocodesGW 199836380 199836257 IN IP4 192.168.161.241 (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=Phone-Call (12) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.241 (24) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 10000 RTP/AVP 8 101 (27) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-15 (15) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (11 headers 11 lines) --- [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #215 [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '6b2e2d1e5a9a08ae3b3c9d8f4d64afdf@pbx.example.net' of Request 102: Match Not Found Really destroying SIP dialog '6b2e2d1e5a9a08ae3b3c9d8f4d64afdf@pbx.example.net' Method: OPTIONS [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: OPTIONS sip:102@192.168.161.241;user=phone SIP/2.0 (50) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1201428a;rport (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as25557bc8 (62) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (40) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (39) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 2d3c2c3163aed5af0a0aefab3568f261@pbx.example.net (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Date: Tue, 11 Sep 2007 08:08:01 GMT (35) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 192.168.161.241:5060: OPTIONS sip:102@192.168.161.241;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1201428a;rport From: "asterisk" ;tag=as25557bc8 To: Contact: Call-ID: 2d3c2c3163aed5af0a0aefab3568f261@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #218 <--- SIP read from 192.168.161.241:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1201428a;rport From: "asterisk" ;tag=as25557bc8 To: ;tag=1c200083268 Call-ID: 2d3c2c3163aed5af0a0aefab3568f261@pbx.example.net CSeq: 102 OPTIONS Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE X-Resources: telchs=4/0;mediachs=0/0 Content-Type: application/sdp Content-Length: 238 v=0 o=AudiocodesGW 200086409 200086285 IN IP4 192.168.161.241 s=Phone-Call c=IN IP4 192.168.161.241 t=0 0 m=audio 10000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 OK (14) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1201428a;rport (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as25557bc8 (62) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=1c200083268 (56) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 2d3c2c3163aed5af0a0aefab3568f261@pbx.example.net (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Supported: em,100rel,timer,replaces,path,resource-priority (58) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: X-Resources: telchs=4/0;mediachs=0/0 (36) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Content-Type: application/sdp (29) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Content-Length: 238 (19) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: (0) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=AudiocodesGW 200086409 200086285 IN IP4 192.168.161.241 (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=Phone-Call (12) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.241 (24) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 10000 RTP/AVP 8 101 (27) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-15 (15) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (11 headers 11 lines) --- [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #218 [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '2d3c2c3163aed5af0a0aefab3568f261@pbx.example.net' of Request 102: Match Not Found Really destroying SIP dialog '2d3c2c3163aed5af0a0aefab3568f261@pbx.example.net' Method: OPTIONS [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: OPTIONS sip:101@192.168.161.241;user=phone SIP/2.0 (50) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK0a3e67ce;rport (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as632e76c7 (62) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (40) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (39) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 25f388312488b29760feb9b24b57b6fd@pbx.example.net (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Date: Tue, 11 Sep 2007 08:08:01 GMT (35) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 192.168.161.241:5060: OPTIONS sip:101@192.168.161.241;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK0a3e67ce;rport From: "asterisk" ;tag=as632e76c7 To: Contact: Call-ID: 25f388312488b29760feb9b24b57b6fd@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #221 <--- SIP read from 192.168.161.241:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK0a3e67ce;rport From: "asterisk" ;tag=as632e76c7 To: ;tag=1c200332470 Call-ID: 25f388312488b29760feb9b24b57b6fd@pbx.example.net CSeq: 102 OPTIONS Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE X-Resources: telchs=4/0;mediachs=0/0 Content-Type: application/sdp Content-Length: 238 v=0 o=AudiocodesGW 200335586 200335461 IN IP4 192.168.161.241 s=Phone-Call c=IN IP4 192.168.161.241 t=0 0 m=audio 10000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 OK (14) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK0a3e67ce;rport (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as632e76c7 (62) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=1c200332470 (56) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 25f388312488b29760feb9b24b57b6fd@pbx.example.net (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Supported: em,100rel,timer,replaces,path,resource-priority (58) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: X-Resources: telchs=4/0;mediachs=0/0 (36) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Content-Type: application/sdp (29) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Content-Length: 238 (19) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: (0) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=AudiocodesGW 200335586 200335461 IN IP4 192.168.161.241 (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=Phone-Call (12) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.241 (24) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 10000 RTP/AVP 8 101 (27) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-15 (15) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (11 headers 11 lines) --- [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #221 [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '25f388312488b29760feb9b24b57b6fd@pbx.example.net' of Request 102: Match Not Found Really destroying SIP dialog '25f388312488b29760feb9b24b57b6fd@pbx.example.net' Method: OPTIONS [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: OPTIONS sip:acMP404@192.168.161.3:5060 SIP/2.0 (46) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK73194e16;rport (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as71762b29 (62) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (36) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (39) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 1ffa90ca213129890fc41a2e76db373d@pbx.example.net (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Date: Tue, 11 Sep 2007 08:08:01 GMT (35) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 192.168.161.3:5060: OPTIONS sip:acMP404@192.168.161.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK73194e16;rport From: "asterisk" ;tag=as71762b29 To: Contact: Call-ID: 1ffa90ca213129890fc41a2e76db373d@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #224 <--- SIP read from 192.168.161.3:5060 ---> SIP/2.0 501 Not Implemented From: "asterisk";tag=as71762b29 To: ;tag=1392f98-0-13c4-2ca41f-6808f83a-2ca41f Call-ID: 1ffa90ca213129890fc41a2e76db373d@pbx.example.net CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.161.100:5060;rport=5060;branch=z9hG4bK73194e16 Supported: replaces,timer,100rel Content-Length: 0 <-------------> [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 501 Not Implemented (27) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: From: "asterisk";tag=as71762b29 (61) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: To: ;tag=1392f98-0-13c4-2ca41f-6808f83a-2ca41f (78) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: Call-ID: 1ffa90ca213129890fc41a2e76db373d@pbx.example.net (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Via: SIP/2.0/UDP 192.168.161.100:5060;rport=5060;branch=z9hG4bK73194e16 (71) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Supported: replaces,timer,100rel (32) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Content-Length: 0 (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #224 [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '1ffa90ca213129890fc41a2e76db373d@pbx.example.net' of Request 102: Match Not Found Really destroying SIP dialog '1ffa90ca213129890fc41a2e76db373d@pbx.example.net' Method: OPTIONS [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: OPTIONS sip:acM1000@192.168.161.2;user=phone SIP/2.0 (52) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK3fc2c2df;rport (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as43b8afef (62) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (42) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (39) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 72f472e508ee27cf2f90fde33b0b7ed7@pbx.example.net (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Date: Tue, 11 Sep 2007 08:08:01 GMT (35) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 192.168.161.2:5060: OPTIONS sip:acM1000@192.168.161.2;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK3fc2c2df;rport From: "asterisk" ;tag=as43b8afef To: Contact: Call-ID: 72f472e508ee27cf2f90fde33b0b7ed7@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #227 <--- SIP read from 192.168.161.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK3fc2c2df;rport From: "asterisk" ;tag=as43b8afef To: ;tag=1c219784641 Call-ID: 72f472e508ee27cf2f90fde33b0b7ed7@pbx.example.net CSeq: 102 OPTIONS Contact: Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 234 X-Resources: telchs=29/29;mediachs=0/0 v=0 o=AudiocodesGW 219797338 219792074 IN IP4 192.168.161.2 s=Phone-Call c=IN IP4 192.168.161.2 t=0 0 m=audio 10000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 OK (14) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK3fc2c2df;rport (66) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as43b8afef (62) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=1c219784641 (58) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 72f472e508ee27cf2f90fde33b0b7ed7@pbx.example.net (57) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: (39) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Supported: em,100rel,timer,replaces,path (40) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Content-Type: application/sdp (29) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Content-Length: 234 (19) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: X-Resources: telchs=29/29;mediachs=0/0 (38) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: (0) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=AudiocodesGW 219797338 219792074 IN IP4 192.168.161.2 (55) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=Phone-Call (12) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.2 (22) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 10000 RTP/AVP 8 101 (27) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-15 (15) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (12 headers 11 lines) --- [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #227 [Sep 11 10:08:01] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '72f472e508ee27cf2f90fde33b0b7ed7@pbx.example.net' of Request 102: Match Not Found Really destroying SIP dialog '72f472e508ee27cf2f90fde33b0b7ed7@pbx.example.net' Method: OPTIONS <--- SIP read from 192.168.161.246:5060 ---> INVITE sip:704@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-kokd8yw0ebyw;rport From: "Rámon" ;tag=pcfg2ibhgy To: ;tag=as5d5ffe6e Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 3 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/6.5.10 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 273 v=0 o=root 1313859764 1313859766 IN IP4 192.168.161.246 s=call c=IN IP4 192.168.161.246 t=0 0 m=audio 14868 RTP/AVP 8 0 3 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendonly <-------------> [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:704@192.168.161.100 SIP/2.0 (38) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-kokd8yw0ebyw;rport (71) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=pcfg2ibhgy (55) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=as5d5ffe6e (55) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 (55) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 3 INVITE (14) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1 (63) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: P-Key-Flags: resolution="31x13", keys="4" (41) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: User-Agent: snom360/6.5.10 (26) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Accept: application/sdp (23) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Allow-Events: talk, hold, refer (31) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: Session-Expires: 3600;refresher=uas (35) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 15: Min-SE: 90 (10) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 16: Content-Type: application/sdp (29) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 17: Content-Length: 273 (19) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 18: (0) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 1313859764 1313859766 IN IP4 192.168.161.246 (51) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.246 (24) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 14868 RTP/AVP 8 0 3 101 (31) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:0 pcmu/8000 (20) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:3 gsm/8000 (19) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendonly (10) --- (18 headers 13 lines) --- [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 192.168.161.246 : 5060 (NAT) Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4993 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 192.168.161.246:14868 Found description format pcma for ID 8 Found description format pcmu for ID 0 Found description format gsm for ID 3 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel SIP/401-09355708 Capabilities: us - 0x8 (alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.246:14868 [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:5313 process_sdp: We have an owner, now see if we need to change this call [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:13601 handle_request_invite: Got a SIP re-invite for call 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:13698 handle_request_invite: SIP/401-09355708: This call is UP.... [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:6499 transmit_response_with_sdp: Setting framing from config on incoming call [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.161.100 port 14096 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) <--- Transmitting (NAT) to 192.168.161.246:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-kokd8yw0ebyw;received=192.168.161.246;rport=5060 From: "Rámon" ;tag=pcfg2ibhgy To: ;tag=as5d5ffe6e Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 3 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18442 IN IP4 192.168.161.247 s=session c=IN IP4 192.168.161.247 t=0 0 m=audio 18596 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:17153 sip_set_rtp_peer: Sending reinvite on SIP '16fb24774565e40c06ce92b510a92129@pbx.example.net' - It's audio soon redirected to IP 192.168.161.100 [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:5713 reqprep: Strict routing enforced for session 16fb24774565e40c06ce92b510a92129@pbx.example.net set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.247, port 5060 [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 192.168.161.100 port 19310 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:1629 initialize_initreq: Initializing already initialized SIP dialog 16fb24774565e40c06ce92b510a92129@pbx.example.net (presumably reinvite) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 (57) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4fab66cd;rport (66) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=as456f0c98 (55) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=viejglm0sj (63) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 4: Contact: (34) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net (57) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 6: CSeq: 104 INVITE (16) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 9: Remote-Party-ID: "Rámon" ;privacy=off;screen=no (73) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 12: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 13: Content-Type: application/sdp (29) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 14: Content-Length: 246 (19) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 15: (0) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: o=root 18440 18442 IN IP4 192.168.161.100 (41) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: s=session (9) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.100 (24) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: m=audio 19310 RTP/AVP 8 101 (27) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=silenceSupp:off - - - - (25) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.161.247:5060: INVITE sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4fab66cd;rport From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Contact: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 104 INVITE User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Rámon" ;privacy=off;screen=no Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18442 IN IP4 192.168.161.100 s=session c=IN IP4 192.168.161.100 t=0 0 m=audio 19310 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 11 10:08:05] DEBUG[18518]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #230  -- Started music on hold, class 'default', on SIP/404-0935a2b8 [Sep 11 10:08:05] DEBUG[18518]: rtp.c:2882 bridge_native_loop: Oooh, 'SIP/401-09355708' changed end address to 0.0.0.0:0 (format 14) [Sep 11 10:08:05] DEBUG[18518]: rtp.c:2884 bridge_native_loop: Oooh, 'SIP/401-09355708' was 192.168.161.246:14868/(format 14) <--- SIP read from 192.168.161.247:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4fab66cd;rport=5060 From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 104 INVITE Contact: ;flow-id=1 User-Agent: snom360/6.5.10 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 224 v=0 o=root 500070299 500070302 IN IP4 192.168.161.247 s=call c=IN IP4 192.168.161.247 t=0 0 m=audio 18596 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 Ok (14) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4fab66cd;rport=5060 (71) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=as456f0c98 (55) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=viejglm0sj (63) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net (57) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 104 INVITE (16) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom360/6.5.10 (26) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow-Events: talk, hold, refer (31) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Type: application/sdp (29) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 224 (19) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 500070299 500070302 IN IP4 192.168.161.247 (49) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.247 (24) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 18596 RTP/AVP 8 101 (27) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (13 headers 11 lines) --- [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:2120 __sip_ack: Acked pending invite 104 [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #230 [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '16fb24774565e40c06ce92b510a92129@pbx.example.net' of Request 104: Match Not Found [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:11737 handle_response_invite: SIP response 200 to RE-invite on outgoing call 16fb24774565e40c06ce92b510a92129@pbx.example.net Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.161.247:18596 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel SIP/404-0935a2b8 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.247:18596 [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:5313 process_sdp: We have an owner, now see if we need to change this call [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:3058 update_call_counter: Updating call counter for outgoing call [Sep 11 10:08:05] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/404 [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:8004 build_route: build_route: Retaining previous route: [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 16fb24774565e40c06ce92b510a92129@pbx.example.net set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.247, port 5060 Transmitting (no NAT) to 192.168.161.247:5060: ACK sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK60bbec4a;rport From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Contact: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 104 ACK User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Rámon" ;privacy=off;screen=no Content-Length: 0 --- [Sep 11 10:08:05] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:08:05] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:08:05] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/404 - state 2 (In use) [Sep 11 10:08:05] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:08:05] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:08:05] DEBUG[18529]: app_queue.c:548 changethread: Device 'SIP/404' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Sep 11 10:08:05] DEBUG[18518]: res_musiconhold.c:261 ast_moh_files_next: SIP/404-0935a2b8 Opened file 1 '/var/lib/asterisk/moh/default/calm-river' [Sep 11 10:08:05] DEBUG[18518]: rtp.c:2597 ast_rtp_raw_write: Difference is 43944, ms is 5513 <--- SIP read from 192.168.161.246:5060 ---> ACK sip:704@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-6zw3a4uubqa2;rport From: "Rámon" ;tag=pcfg2ibhgy To: ;tag=as5d5ffe6e Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 3 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: ACK sip:704@192.168.161.100 SIP/2.0 (35) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-6zw3a4uubqa2;rport (71) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=pcfg2ibhgy (55) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=as5d5ffe6e (55) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 (55) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 3 ACK (11) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1 (63) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Content-Length: 0 (17) [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received ACK (6) - Command in SIP ACK [Sep 11 10:08:05] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '3c2681f30271-phvhk1y47mv2@snom360-000413231BC4' of Response 3: Match Found <--- SIP read from 192.168.161.246:5060 ---> REFER sip:403@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-yruchw2ll6yt;rport From: ;tag=ull0udxr9r To: "Theo Belder" ;tag=as1b4cb1fd Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 2 REFER Max-Forwards: 70 Contact: ;flow-id=1 Refer-To: sip:704@192.168.161.100?Replaces=3c2681f30271-phvhk1y47mv2%40snom360-000413231BC4%3Bto-tag%3Das5d5ffe6e%3Bfrom-tag%3Dpcfg2ibhgy Referred-By: sip:401@pbx.example.net User-Agent: snom360/6.5.10 Content-Length: 0 <-------------> [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REFER sip:403@192.168.161.100 SIP/2.0 (37) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-yruchw2ll6yt;rport (71) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: ;tag=ull0udxr9r (65) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Theo Belder" ;tag=as1b4cb1fd (58) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net (57) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 2 REFER (13) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1 (63) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Refer-To: sip:704@192.168.161.100?Replaces=3c2681f30271-phvhk1y47mv2%40snom360-000413231BC4%3Bto-tag%3Das5d5ffe6e%3Bfrom-tag%3Dpcfg2ibhgy (137) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Referred-By: sip:401@pbx.example.net (36) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: User-Agent: snom360/6.5.10 (26) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Length: 0 (17) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REFER (9) - Command in SIP REFER Call 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net got a SIP call transfer from caller: (REFER)! [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:8790 get_refer_info: Attended transfer: Will use Replace-Call-ID : 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 (No check of from/to tags) SIP transfer to extension 704@user-01 by 401@pbx.example.net [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:14066 handle_request_refer: This SIP transfer is to a remote SIP extension (remote domain 192.168.161.100) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:14105 handle_request_refer: SIP attended transfer: Transferer channel SIP/401-0933a850, transferee channel SIP/403-09334c20 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:14121 handle_request_refer: Got SIP transfer, applying to bridged peer 'SIP/403-09334c20' [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:8644 get_sip_pvt_byid_locked: Looking for callid 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 (fromtag pcfg2ibhgy totag as5d5ffe6e) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:8668 get_sip_pvt_byid_locked: Matched INCOMING call - their tag is pcfg2ibhgy Our tag is as5d5ffe6e <--- Transmitting (NAT) to 192.168.161.246:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-yruchw2ll6yt;received=192.168.161.246;rport=5060 From: ;tag=ull0udxr9r To: "Theo Belder" ;tag=as1b4cb1fd Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 2 REFER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:13873 local_attended_transfer: SIP attended transfer: trying to bridge SIP/401-09355708 and SIP/403-09334c20 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:12875 attempt_transfer: Sip transfer:-------------------- [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:12877 attempt_transfer: -- Transferer to PBX channel: SIP/401-0933a850 State Up [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:12881 attempt_transfer: -- Transferer to PBX second channel (target): SIP/401-09355708 State Up [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:12885 attempt_transfer: -- Bridged call to transferee: SIP/403-09334c20 State Up [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:12889 attempt_transfer: -- Bridged call to transfer target: SIP/404-0935a2b8 State Up [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:12892 attempt_transfer: -- END Sip transfer:-------------------- [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:12900 attempt_transfer: SIP transfer: Four channels to handle  -- Stopped music on hold on SIP/403-09334c20  -- Stopped music on hold on SIP/404-0935a2b8 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:12931 attempt_transfer: SIP transfer: trying to masquerade SIP/403-09334c20 into SIP/401-09355708 [Sep 11 10:08:06] DEBUG[18453]: channel.c:3447 ast_channel_masquerade: Planning to masquerade channel SIP/403-09334c20 into the structure of SIP/401-09355708 [Sep 11 10:08:06] DEBUG[18453]: channel.c:3461 ast_channel_masquerade: Done planning to masquerade channel SIP/403-09334c20 into the structure of SIP/401-09355708 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:12936 attempt_transfer: SIP transfer: Succeeded to masquerade channels. [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.246, port 5060 Reliably Transmitting (NAT) to 192.168.161.246:5060: NOTIFY sip:401@192.168.161.246:5060;line=4f6xat3o SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2d917ae8;rport From: "Theo Belder" ;tag=as1b4cb1fd To: ;tag=ull0udxr9r Contact: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 104 NOTIFY User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Theo Belder" ;privacy=off;screen=no Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 16 SIP/2.0 200 OK --- [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #231 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:13904 local_attended_transfer: SIP attended transfer: Unlocking channel SIP/401-09355708 [Sep 11 10:08:06] DEBUG[18518]: channel.c:3573 ast_do_masquerade: Actually Masquerading SIP/403-09334c20(6) into the structure of SIP/401-09355708(6) [Sep 11 10:08:06] DEBUG[18518]: channel.c:3585 ast_do_masquerade: Got clone lock for masquerade on 'SIP/403-09334c20' at 0x9339de8 [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:3650 sip_fixup: SIP Fixup: New owner for dialogue 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4: SIP/403-09334c20 (Old parent: SIP/403-09334c20) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:3368 sip_hangup: Hangup call SIP/403-09334c20, SIP callid 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:3377 sip_hangup: update_call_counter(401) - decrement call limit counter on hangup [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:3058 update_call_counter: Updating call counter for incoming call [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:3107 update_call_counter: Call from peer '401' removed from call limit 30 [Sep 11 10:08:06] DEBUG[18518]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/401 Scheduling destruction of SIP dialog '3c2681f30271-phvhk1y47mv2@snom360-000413231BC4' in 32000 ms (Method: ACK) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:5713 reqprep: Strict routing enforced for session 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.246, port 5060 Reliably Transmitting (NAT) to 192.168.161.246:5060: BYE sip:401@192.168.161.246:5060;line=4f6xat3o SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4d1ac348;rport From: ;tag=as5d5ffe6e To: "Rámon" ;tag=pcfg2ibhgy Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 103 BYE User-Agent: atCOM PBX Max-Forwards: 70 Content-Length: 0 --- [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #233 [Sep 11 10:08:06] DEBUG[18518]: channel.c:3786 ast_do_masquerade: Putting channel SIP/403-09334c20 in 8/8 formats [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:3650 sip_fixup: SIP Fixup: New owner for dialogue 3c26823294ed-t2wihiydodry@snom320-0004132496DC: SIP/403-09334c20 (Old parent: SIP/401-09355708) [Sep 11 10:08:06] DEBUG[18518]: channel.c:3822 ast_do_masquerade: Released clone lock on 'SIP/401-09355708' [Sep 11 10:08:06] DEBUG[18518]: channel.c:3832 ast_do_masquerade: Done Masquerading SIP/403-09334c20 (6) [Sep 11 10:08:06] DEBUG[18518]: rtp.c:2839 bridge_native_loop: Oooh, something is weird, backing out  -- Native bridging SIP/403-09334c20 and SIP/404-0935a2b8 [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:17153 sip_set_rtp_peer: Sending reinvite on SIP '3c26823294ed-t2wihiydodry@snom320-0004132496DC' - It's audio soon redirected to IP 192.168.161.247 [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:5713 reqprep: Strict routing enforced for session 3c26823294ed-t2wihiydodry@snom320-0004132496DC set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.250, port 5060 [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.161.100 port 17674 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:1629 initialize_initreq: Initializing already initialized SIP dialog 3c26823294ed-t2wihiydodry@snom320-0004132496DC (presumably reinvite) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:403@192.168.161.250:5060;line=xjqx46za SIP/2.0 (57) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK07385df8;rport (66) [Sep 11 10:08:06] DEBUG[18501]: rtp.c:2839 bridge_native_loop: Oooh, something is weird, backing out [Sep 11 10:08:06] DEBUG[18501]: channel.c:4233 ast_channel_bridge: Bridge stops because we're zombie or need a soft hangup: c0=SIP/401-09355708, c1=SIP/401-0933a850, flags: Yes,Yes,No,No [Sep 11 10:08:06] DEBUG[18501]: channel.c:4338 ast_channel_bridge: Bridge stops bridging channels SIP/401-09355708 and SIP/401-0933a850 [Sep 11 10:08:06] DEBUG[18501]: channel.c:1764 ast_hangup: Hanging up channel 'SIP/401-0933a850' [Sep 11 10:08:06] DEBUG[18501]: chan_sip.c:3349 sip_hangup: update_call_counter(401) - decrement call limit counter on hangup [Sep 11 10:08:06] DEBUG[18501]: chan_sip.c:3058 update_call_counter: Updating call counter for outgoing call [Sep 11 10:08:06] DEBUG[18501]: chan_sip.c:3107 update_call_counter: Call to peer '401' removed from call limit 30 [Sep 11 10:08:06] DEBUG[18501]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/401 [Sep 11 10:08:06] DEBUG[18501]: chan_sip.c:3353 sip_hangup: SIP Transfer: Not hanging up right now... Rescheduling hangup for 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net. Scheduling destruction of SIP dialog '0e3f4f4e700219793dcc419c77a3b373@pbx.example.net' in 32000 ms (Method: REFER) [Sep 11 10:08:06] DEBUG[18501]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/401-0933a850 [Sep 11 10:08:06] DEBUG[18501]: rtp.c:1499 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Sep 11 10:08:06] DEBUG[18501]: app_dial.c:1686 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Sep 11 10:08:06] DEBUG[18501]: pbx.c:2407 __ast_pbx_run: Spawn extension (user-01,701,1) exited non-zero on 'SIP/401-09355708'  == Spawn extension (user-01, 701, 1) exited non-zero on 'SIP/401-09355708' [Sep 11 10:08:06] DEBUG[18501]: channel.c:1547 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/401-09355708' [Sep 11 10:08:06] DEBUG[18501]: channel.c:1769 ast_hangup: Hanging up zombie 'SIP/401-09355708' [Sep 11 10:08:06] DEBUG[18501]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/401-09355708 [Sep 11 10:08:06] DEBUG[18501]: cdr.c:1164 ast_cdr_detach: Dropping CDR ! [Sep 11 10:08:06] NOTICE[18501]: cdr.c:432 ast_cdr_free: CDR on channel 'SIP/401-0933a850' not posted [Sep 11 10:08:06] NOTICE[18501]: cdr.c:432 ast_cdr_free: CDR on channel 'SIP/403-09334c20' not posted [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 2: From: ;tag=as449bf3fa (57) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 3: To: "Theo Belder" ;tag=bjxxjlf0hu (58) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 4: Contact: (34) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC (55) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 6: CSeq: 105 INVITE (16) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 10: Supported: replaces (19) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 12: Content-Type: application/sdp (29) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 13: Content-Length: 246 (19) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 14: (0) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: o=root 18440 18444 IN IP4 192.168.161.247 (41) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: s=session (9) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.247 (24) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: m=audio 18596 RTP/AVP 8 101 (27) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=silenceSupp:off - - - - (25) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.161.250:5060: INVITE sip:403@192.168.161.250:5060;line=xjqx46za SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK07385df8;rport From: ;tag=as449bf3fa To: "Theo Belder" ;tag=bjxxjlf0hu Contact: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 105 INVITE User-Agent: atCOM PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18444 IN IP4 192.168.161.247 s=session c=IN IP4 192.168.161.247 t=0 0 m=audio 18596 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #235 [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:17153 sip_set_rtp_peer: Sending reinvite on SIP '16fb24774565e40c06ce92b510a92129@pbx.example.net' - It's audio soon redirected to IP 192.168.161.250 [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:5713 reqprep: Strict routing enforced for session 16fb24774565e40c06ce92b510a92129@pbx.example.net set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.247, port 5060 [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 192.168.161.100 port 19310 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:1629 initialize_initreq: Initializing already initialized SIP dialog 16fb24774565e40c06ce92b510a92129@pbx.example.net (presumably reinvite) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 (57) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK33e972bd;rport (66) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=as456f0c98 (55) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=viejglm0sj (63) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 4: Contact: (34) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net (57) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 6: CSeq: 105 INVITE (16) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 9: Remote-Party-ID: "Rámon" ;privacy=off;screen=no (73) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 12: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 13: Content-Type: application/sdp (29) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 14: Content-Length: 246 (19) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 15: (0) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: o=root 18440 18443 IN IP4 192.168.161.250 (41) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: s=session (9) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.250 (24) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: m=audio 15084 RTP/AVP 8 101 (27) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=silenceSupp:off - - - - (25) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.161.247:5060: INVITE sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK33e972bd;rport From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Contact: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 105 INVITE User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Rámon" ;privacy=off;screen=no Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18443 IN IP4 192.168.161.250 s=session c=IN IP4 192.168.161.250 t=0 0 m=audio 15084 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 11 10:08:06] DEBUG[18518]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #236 [Sep 11 10:08:06] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/401 - state 1 (Not in use) [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:08:06] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:08:06] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/401 - state 1 (Not in use) [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:08:06] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:08:06] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/401 - state 1 (Not in use) [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:08:06] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:08:06] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/401 - state 1 (Not in use) [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:08:06] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:08:06] DEBUG[18530]: app_queue.c:548 changethread: Device 'SIP/401' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 11 10:08:06] DEBUG[18531]: app_queue.c:548 changethread: Device 'SIP/401' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 11 10:08:06] DEBUG[18532]: app_queue.c:548 changethread: Device 'SIP/401' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 11 10:08:06] DEBUG[18533]: app_queue.c:548 changethread: Device 'SIP/401' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.247:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK33e972bd;rport=5060 From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 105 INVITE Contact: ;flow-id=1 User-Agent: snom360/6.5.10 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 224 v=0 o=root 500070299 500070303 IN IP4 192.168.161.247 s=call c=IN IP4 192.168.161.247 t=0 0 m=audio 18596 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 Ok (14) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK33e972bd;rport=5060 (71) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=as456f0c98 (55) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=viejglm0sj (63) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net (57) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 105 INVITE (16) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom360/6.5.10 (26) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow-Events: talk, hold, refer (31) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Type: application/sdp (29) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 224 (19) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 500070299 500070303 IN IP4 192.168.161.247 (49) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.247 (24) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 18596 RTP/AVP 8 101 (27) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (13 headers 11 lines) --- [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:2120 __sip_ack: Acked pending invite 105 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #236 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '16fb24774565e40c06ce92b510a92129@pbx.example.net' of Request 105: Match Not Found [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:11737 handle_response_invite: SIP response 200 to RE-invite on outgoing call 16fb24774565e40c06ce92b510a92129@pbx.example.net Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.161.247:18596 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel SIP/404-0935a2b8 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.247:18596 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:5313 process_sdp: We have an owner, now see if we need to change this call [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:3058 update_call_counter: Updating call counter for outgoing call [Sep 11 10:08:06] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/404 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:8004 build_route: build_route: Retaining previous route: [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 16fb24774565e40c06ce92b510a92129@pbx.example.net set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.247, port 5060 Transmitting (no NAT) to 192.168.161.247:5060: ACK sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK7c424f6e;rport From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Contact: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 105 ACK User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Rámon" ;privacy=off;screen=no Content-Length: 0 --- [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:08:06] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/404 - state 2 (In use) [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:08:06] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:08:06] DEBUG[18534]: app_queue.c:548 changethread: Device 'SIP/404' changed to state '2' (In use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.250:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK07385df8;rport=5060 From: ;tag=as449bf3fa To: "Theo Belder" ;tag=bjxxjlf0hu Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 105 INVITE Contact: ;flow-id=1 User-Agent: snom320/6.5.10 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 224 v=0 o=root 490570670 490570674 IN IP4 192.168.161.250 s=call c=IN IP4 192.168.161.250 t=0 0 m=audio 15084 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 Ok (14) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK07385df8;rport=5060 (71) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: ;tag=as449bf3fa (57) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Theo Belder" ;tag=bjxxjlf0hu (58) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC (55) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 105 INVITE (16) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom320/6.5.10 (26) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow-Events: talk, hold, refer (31) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Type: application/sdp (29) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 224 (19) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 490570670 490570674 IN IP4 192.168.161.250 (49) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.250 (24) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 15084 RTP/AVP 8 101 (27) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (13 headers 11 lines) --- [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:2120 __sip_ack: Acked pending invite 105 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #235 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '3c26823294ed-t2wihiydodry@snom320-0004132496DC' of Request 105: Match Not Found [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:11737 handle_response_invite: SIP response 200 to RE-invite on outgoing call 3c26823294ed-t2wihiydodry@snom320-0004132496DC Found RTP audio format 8 Found RTP audio format 101 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4993 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 192.168.161.250:15084 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel SIP/403-09334c20 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.250:15084 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:5313 process_sdp: We have an owner, now see if we need to change this call [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:3058 update_call_counter: Updating call counter for incoming call [Sep 11 10:08:06] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/403 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:8004 build_route: build_route: Retaining previous route: [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:11870 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:11875 handle_response_invite: T38 state changed to 0 on channel SIP [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:11878 handle_response_invite: T38 state changed to 0 on channel SIP/403-09334c20 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 3c26823294ed-t2wihiydodry@snom320-0004132496DC set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.250, port 5060 Transmitting (no NAT) to 192.168.161.250:5060: ACK sip:403@192.168.161.250:5060;line=xjqx46za SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK5acbe2cd;rport From: ;tag=as449bf3fa To: "Theo Belder" ;tag=bjxxjlf0hu Contact: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 105 ACK User-Agent: atCOM PBX Max-Forwards: 70 Content-Length: 0 --- [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:08:06] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/403 - state 2 (In use) [Sep 11 10:08:06] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:08:06] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:08:06] DEBUG[18535]: app_queue.c:548 changethread: Device 'SIP/403' changed to state '2' (In use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.246:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2d917ae8;rport=5060 From: "Theo Belder" ;tag=as1b4cb1fd To: ;tag=ull0udxr9r Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 104 NOTIFY Contact: ;flow-id=1 Content-Length: 0 <-------------> [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 Ok (14) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2d917ae8;rport=5060 (71) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Theo Belder" ;tag=as1b4cb1fd (60) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=ull0udxr9r (63) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net (57) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 104 NOTIFY (16) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Content-Length: 0 (17) [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #231 [Sep 11 10:08:06] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '0e3f4f4e700219793dcc419c77a3b373@pbx.example.net' of Request 104: Match Not Found SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 192.168.161.246:5060 ---> BYE sip:403@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-oe276gjnd59y;rport From: ;tag=ull0udxr9r To: "Theo Belder" ;tag=as1b4cb1fd Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 3 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/6.5.10 RTP-RxStat: Total_Rx_Pkts=158,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=165,Tx_Pkts=0,Remote_Tx_Pkts=0 Content-Length: 0 <-------------> [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: BYE sip:403@192.168.161.100 SIP/2.0 (35) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-oe276gjnd59y;rport (71) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: ;tag=ull0udxr9r (65) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Theo Belder" ;tag=as1b4cb1fd (58) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net (57) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 3 BYE (11) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1 (63) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: User-Agent: snom360/6.5.10 (26) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: RTP-RxStat: Total_Rx_Pkts=158,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 (76) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: RTP-TxStat: Total_Tx_Pkts=165,Tx_Pkts=0,Remote_Tx_Pkts=0 (56) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Length: 0 (17) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 192.168.161.246 : 5060 (NAT) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:1641 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net Scheduling destruction of SIP dialog '0e3f4f4e700219793dcc419c77a3b373@pbx.example.net' in 32000 ms (Method: BYE) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:14411 handle_request_bye: Received bye, no owner, selfdestruct soon. <--- Transmitting (NAT) to 192.168.161.246:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-oe276gjnd59y;received=192.168.161.246;rport=5060 From: ;tag=ull0udxr9r To: "Theo Belder" ;tag=as1b4cb1fd Call-ID: 0e3f4f4e700219793dcc419c77a3b373@pbx.example.net CSeq: 3 BYE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- SIP read from 192.168.161.246:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4d1ac348;rport=5060 From: ;tag=as5d5ffe6e To: "Rámon" ;tag=pcfg2ibhgy Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 CSeq: 103 BYE Contact: ;flow-id=1 User-Agent: snom360/6.5.10 RTP-RxStat: Total_Rx_Pkts=259,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=272,Tx_Pkts=0,Remote_Tx_Pkts=0 Content-Length: 0 <-------------> [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 OK (14) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4d1ac348;rport=5060 (71) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: ;tag=as5d5ffe6e (57) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Rámon" ;tag=pcfg2ibhgy (53) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 (55) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 103 BYE (13) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom360/6.5.10 (26) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: RTP-RxStat: Total_Rx_Pkts=259,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 (76) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: RTP-TxStat: Total_Tx_Pkts=272,Tx_Pkts=0,Remote_Tx_Pkts=0 (56) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Content-Length: 0 (17) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: (0) --- (11 headers 0 lines) --- [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #233 [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '3c2681f30271-phvhk1y47mv2@snom360-000413231BC4' of Request 103: Match Not Found SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '3c2681f30271-phvhk1y47mv2@snom360-000413231BC4' Method: ACK [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:3058 update_call_counter: Updating call counter for incoming call [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:3107 update_call_counter: Call from peer '401' removed from call limit 30 [Sep 11 10:08:07] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/401 [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:2949 __sip_destroy: This call did not properly clean up call limits. Call ID 3c2681f30271-phvhk1y47mv2@snom360-000413231BC4 [Sep 11 10:08:07] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:08:07] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:08:07] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/401 - state 1 (Not in use) [Sep 11 10:08:07] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:08:07] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:08:07] DEBUG[18536]: app_queue.c:548 changethread: Device 'SIP/401' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: OPTIONS sip:192.168.1.160 SIP/2.0 (33) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4de91f1c;rport (66) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as226d67d7 (62) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (23) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (39) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 74f02c71581ea338096409fd0a1abc59@pbx.example.net (57) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Date: Tue, 11 Sep 2007 08:08:07 GMT (35) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4de91f1c;rport From: "asterisk" ;tag=as226d67d7 To: Contact: Call-ID: 74f02c71581ea338096409fd0a1abc59@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:07] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #238 [Sep 11 10:08:08] DEBUG[18453]: chan_sip.c:1879 retrans_pkt: SIP TIMER: Not rescheduling id #238:OPTIONS (Method 3) (No timer T1) Retransmitting #1 (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4de91f1c;rport From: "asterisk" ;tag=as226d67d7 To: Contact: Call-ID: 74f02c71581ea338096409fd0a1abc59@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 192.168.161.241:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac219576184 Max-Forwards: 70 From: ;tag=1c219573317 To: Call-ID: 172448606018200722437@192.168.161.241 CSeq: 232352 REGISTER Contact: ;expires=60 Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 60 User-Agent: acMP-114/v.5.00A.043 Content-Length: 0 <-------------> [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac219576184 (58) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: Max-Forwards: 70 (16) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: From: ;tag=1c219573317 (58) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: To: (40) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 172448606018200722437@192.168.161.241 (46) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 232352 REGISTER (21) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;expires=60 (56) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Supported: em,timer,replaces,path,resource-priority (51) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Expires: 60 (11) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: User-Agent: acMP-114/v.5.00A.043 (32) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) --- (13 headers 0 lines) --- [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.241 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac219576184;received=192.168.161.241 From: ;tag=1c219573317 To: Call-ID: 172448606018200722437@192.168.161.241 CSeq: 232352 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac219576184;received=192.168.161.241 From: ;tag=1c219573317 To: ;tag=as41885f68 Call-ID: 172448606018200722437@192.168.161.241 CSeq: 232352 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="siprealm", nonce="0fd80feb" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '172448606018200722437@192.168.161.241' in 32000 ms (Method: REGISTER) <--- SIP read from 192.168.161.241:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac219657735 Max-Forwards: 70 From: ;tag=1c219573317 To: Call-ID: 172448606018200722437@192.168.161.241 CSeq: 232353 REGISTER Authorization: Digest username="102",realm="siprealm",nonce="0fd80feb",uri="sip:pbx.example.net",algorithm=MD5,response="006cd4a8d73600ac3556617eefda4c75" Contact: ;expires=60 Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 60 User-Agent: acMP-114/v.5.00A.043 Content-Length: 0 <-------------> [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac219657735 (58) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: Max-Forwards: 70 (16) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: From: ;tag=1c219573317 (58) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: To: (40) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 172448606018200722437@192.168.161.241 (46) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 232353 REGISTER (21) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Authorization: Digest username="102",realm="siprealm",nonce="0fd80feb",uri="sip:pbx.example.net",algorithm=MD5,response="006cd4a8d73600ac3556617eefda4c75" (154) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Contact: ;expires=60 (56) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Supported: em,timer,replaces,path,resource-priority (51) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Expires: 60 (11) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: User-Agent: acMP-114/v.5.00A.043 (32) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Content-Length: 0 (17) [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: (0) --- (14 headers 0 lines) --- [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.241 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac219657735;received=192.168.161.241 From: ;tag=1c219573317 To: Call-ID: 172448606018200722437@192.168.161.241 CSeq: 232353 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac219657735;received=192.168.161.241 From: ;tag=1c219573317 To: ;tag=as41885f68 Call-ID: 172448606018200722437@192.168.161.241 CSeq: 232353 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Tue, 11 Sep 2007 08:08:09 GMT Content-Length: 0 <------------> [Sep 11 10:08:09] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/102 Scheduling destruction of SIP dialog '172448606018200722437@192.168.161.241' in 32000 ms (Method: REGISTER) [Sep 11 10:08:09] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 102 [Sep 11 10:08:09] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 102 [Sep 11 10:08:09] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/102 - state 1 (Not in use) [Sep 11 10:08:09] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 102 [Sep 11 10:08:09] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 102 [Sep 11 10:08:09] DEBUG[18537]: app_queue.c:548 changethread: Device 'SIP/102' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 11 10:08:09] DEBUG[18453]: chan_sip.c:1879 retrans_pkt: SIP TIMER: Not rescheduling id #238:OPTIONS (Method 3) (No timer T1) Retransmitting #2 (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4de91f1c;rport From: "asterisk" ;tag=as226d67d7 To: Contact: Call-ID: 74f02c71581ea338096409fd0a1abc59@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 192.168.161.241:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222575155 Max-Forwards: 70 From: ;tag=1c222572304 To: Call-ID: 172448489118200722437@192.168.161.241 CSeq: 232357 REGISTER Contact: ;expires=60 Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 60 User-Agent: acMP-114/v.5.00A.043 Content-Length: 0 <-------------> [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222575155 (58) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: Max-Forwards: 70 (16) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: From: ;tag=1c222572304 (58) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: To: (40) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 172448489118200722437@192.168.161.241 (46) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 232357 REGISTER (21) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;expires=60 (56) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Supported: em,timer,replaces,path,resource-priority (51) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Expires: 60 (11) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: User-Agent: acMP-114/v.5.00A.043 (32) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) --- (13 headers 0 lines) --- [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.241 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222575155;received=192.168.161.241 From: ;tag=1c222572304 To: Call-ID: 172448489118200722437@192.168.161.241 CSeq: 232357 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222575155;received=192.168.161.241 From: ;tag=1c222572304 To: ;tag=as7f14bce6 Call-ID: 172448489118200722437@192.168.161.241 CSeq: 232357 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="siprealm", nonce="35cb981c" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '172448489118200722437@192.168.161.241' in 32000 ms (Method: REGISTER) <--- SIP read from 192.168.161.241:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222641585 Max-Forwards: 70 From: ;tag=1c222638733 To: Call-ID: 172448548918200722437@192.168.161.241 CSeq: 232354 REGISTER Contact: ;expires=60 Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 60 User-Agent: acMP-114/v.5.00A.043 Content-Length: 0 <-------------> [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222641585 (58) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: Max-Forwards: 70 (16) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: From: ;tag=1c222638733 (58) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: To: (40) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 172448548918200722437@192.168.161.241 (46) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 232354 REGISTER (21) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;expires=60 (56) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Supported: em,timer,replaces,path,resource-priority (51) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Expires: 60 (11) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: User-Agent: acMP-114/v.5.00A.043 (32) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) --- (13 headers 0 lines) --- [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.241 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222641585;received=192.168.161.241 From: ;tag=1c222638733 To: Call-ID: 172448548918200722437@192.168.161.241 CSeq: 232354 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222641585;received=192.168.161.241 From: ;tag=1c222638733 To: ;tag=as26770ef9 Call-ID: 172448548918200722437@192.168.161.241 CSeq: 232354 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="siprealm", nonce="5a544ab3" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '172448548918200722437@192.168.161.241' in 32000 ms (Method: REGISTER) <--- SIP read from 192.168.161.241:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222664559 Max-Forwards: 70 From: ;tag=1c222658777 To: Call-ID: 172448662918200722437@192.168.161.241 CSeq: 232354 REGISTER Contact: ;expires=60 Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 60 User-Agent: acMP-114/v.5.00A.043 Content-Length: 0 <-------------> [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222664559 (58) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: Max-Forwards: 70 (16) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: From: ;tag=1c222658777 (58) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: To: (40) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 172448662918200722437@192.168.161.241 (46) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 232354 REGISTER (21) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;expires=60 (56) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Supported: em,timer,replaces,path,resource-priority (51) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Expires: 60 (11) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: User-Agent: acMP-114/v.5.00A.043 (32) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) --- (13 headers 0 lines) --- [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.241 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222664559;received=192.168.161.241 From: ;tag=1c222658777 To: Call-ID: 172448662918200722437@192.168.161.241 CSeq: 232354 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222664559;received=192.168.161.241 From: ;tag=1c222658777 To: ;tag=as76596d9f Call-ID: 172448662918200722437@192.168.161.241 CSeq: 232354 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="siprealm", nonce="3b530466" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '172448662918200722437@192.168.161.241' in 32000 ms (Method: REGISTER) <--- SIP read from 192.168.161.241:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222686649 Max-Forwards: 70 From: ;tag=1c222572304 To: Call-ID: 172448489118200722437@192.168.161.241 CSeq: 232358 REGISTER Authorization: Digest username="104",realm="siprealm",nonce="35cb981c",uri="sip:pbx.example.net",algorithm=MD5,response="edf308c13e22b87605eb0f1764b06f47" Contact: ;expires=60 Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 60 User-Agent: acMP-114/v.5.00A.043 Content-Length: 0 <-------------> [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222686649 (58) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: Max-Forwards: 70 (16) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: From: ;tag=1c222572304 (58) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: To: (40) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 172448489118200722437@192.168.161.241 (46) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 232358 REGISTER (21) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Authorization: Digest username="104",realm="siprealm",nonce="35cb981c",uri="sip:pbx.example.net",algorithm=MD5,response="edf308c13e22b87605eb0f1764b06f47" (154) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Contact: ;expires=60 (56) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Supported: em,timer,replaces,path,resource-priority (51) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Expires: 60 (11) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: User-Agent: acMP-114/v.5.00A.043 (32) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Content-Length: 0 (17) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: (0) --- (14 headers 0 lines) --- [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.241 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222686649;received=192.168.161.241 From: ;tag=1c222572304 To: Call-ID: 172448489118200722437@192.168.161.241 CSeq: 232358 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222686649;received=192.168.161.241 From: ;tag=1c222572304 To: ;tag=as7f14bce6 Call-ID: 172448489118200722437@192.168.161.241 CSeq: 232358 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Tue, 11 Sep 2007 08:08:10 GMT Content-Length: 0 <------------> [Sep 11 10:08:10] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/104 Scheduling destruction of SIP dialog '172448489118200722437@192.168.161.241' in 32000 ms (Method: REGISTER) [Sep 11 10:08:10] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 104 [Sep 11 10:08:10] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 104 [Sep 11 10:08:10] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/104 - state 1 (Not in use) [Sep 11 10:08:10] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 104 [Sep 11 10:08:10] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 104 [Sep 11 10:08:10] DEBUG[18538]: app_queue.c:548 changethread: Device 'SIP/104' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.241:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222718849 Max-Forwards: 70 From: ;tag=1c222638733 To: Call-ID: 172448548918200722437@192.168.161.241 CSeq: 232355 REGISTER Authorization: Digest username="103",realm="siprealm",nonce="5a544ab3",uri="sip:pbx.example.net",algorithm=MD5,response="33db5367a36e6539cf8547043cca0ff4" Contact: ;expires=60 Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 60 User-Agent: acMP-114/v.5.00A.043 Content-Length: 0 <-------------> [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222718849 (58) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: Max-Forwards: 70 (16) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: From: ;tag=1c222638733 (58) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: To: (40) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 172448548918200722437@192.168.161.241 (46) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 232355 REGISTER (21) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Authorization: Digest username="103",realm="siprealm",nonce="5a544ab3",uri="sip:pbx.example.net",algorithm=MD5,response="33db5367a36e6539cf8547043cca0ff4" (154) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Contact: ;expires=60 (56) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Supported: em,timer,replaces,path,resource-priority (51) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Expires: 60 (11) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: User-Agent: acMP-114/v.5.00A.043 (32) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Content-Length: 0 (17) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: (0) --- (14 headers 0 lines) --- [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.241 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222718849;received=192.168.161.241 From: ;tag=1c222638733 To: Call-ID: 172448548918200722437@192.168.161.241 CSeq: 232355 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222718849;received=192.168.161.241 From: ;tag=1c222638733 To: ;tag=as26770ef9 Call-ID: 172448548918200722437@192.168.161.241 CSeq: 232355 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Tue, 11 Sep 2007 08:08:10 GMT Content-Length: 0 <------------> [Sep 11 10:08:10] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/103 Scheduling destruction of SIP dialog '172448548918200722437@192.168.161.241' in 32000 ms (Method: REGISTER) [Sep 11 10:08:10] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 103 [Sep 11 10:08:10] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 103 [Sep 11 10:08:10] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/103 - state 1 (Not in use) [Sep 11 10:08:10] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 103 [Sep 11 10:08:10] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 103 [Sep 11 10:08:10] DEBUG[18539]: app_queue.c:548 changethread: Device 'SIP/103' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.241:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222750691 Max-Forwards: 70 From: ;tag=1c222658777 To: Call-ID: 172448662918200722437@192.168.161.241 CSeq: 232355 REGISTER Authorization: Digest username="101",realm="siprealm",nonce="3b530466",uri="sip:pbx.example.net",algorithm=MD5,response="59ff19452da00ecd3cbaaa589e335482" Contact: ;expires=60 Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 60 User-Agent: acMP-114/v.5.00A.043 Content-Length: 0 <-------------> [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222750691 (58) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: Max-Forwards: 70 (16) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: From: ;tag=1c222658777 (58) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: To: (40) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 172448662918200722437@192.168.161.241 (46) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 232355 REGISTER (21) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Authorization: Digest username="101",realm="siprealm",nonce="3b530466",uri="sip:pbx.example.net",algorithm=MD5,response="59ff19452da00ecd3cbaaa589e335482" (154) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Contact: ;expires=60 (56) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Supported: em,timer,replaces,path,resource-priority (51) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Expires: 60 (11) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: User-Agent: acMP-114/v.5.00A.043 (32) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: Content-Length: 0 (17) [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: (0) --- (14 headers 0 lines) --- [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.241 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222750691;received=192.168.161.241 From: ;tag=1c222658777 To: Call-ID: 172448662918200722437@192.168.161.241 CSeq: 232355 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.161.241:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.241;branch=z9hG4bKac222750691;received=192.168.161.241 From: ;tag=1c222658777 To: ;tag=as76596d9f Call-ID: 172448662918200722437@192.168.161.241 CSeq: 232355 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Tue, 11 Sep 2007 08:08:10 GMT Content-Length: 0 <------------> [Sep 11 10:08:10] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/101 Scheduling destruction of SIP dialog '172448662918200722437@192.168.161.241' in 32000 ms (Method: REGISTER) [Sep 11 10:08:10] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 101 [Sep 11 10:08:10] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 101 [Sep 11 10:08:10] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/101 - state 1 (Not in use) [Sep 11 10:08:10] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 101 [Sep 11 10:08:10] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 101 [Sep 11 10:08:10] DEBUG[18540]: app_queue.c:548 changethread: Device 'SIP/101' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 11 10:08:10] DEBUG[18453]: chan_sip.c:1879 retrans_pkt: SIP TIMER: Not rescheduling id #238:OPTIONS (Method 3) (No timer T1) Retransmitting #3 (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4de91f1c;rport From: "asterisk" ;tag=as226d67d7 To: Contact: Call-ID: 74f02c71581ea338096409fd0a1abc59@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:11] DEBUG[18453]: chan_sip.c:1879 retrans_pkt: SIP TIMER: Not rescheduling id #238:OPTIONS (Method 3) (No timer T1) Retransmitting #4 (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4de91f1c;rport From: "asterisk" ;tag=as226d67d7 To: Contact: Call-ID: 74f02c71581ea338096409fd0a1abc59@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:11] DEBUG[18453]: chan_sip.c:3165 sip_destroy: Destroying SIP dialog 74f02c71581ea338096409fd0a1abc59@pbx.example.net Really destroying SIP dialog '74f02c71581ea338096409fd0a1abc59@pbx.example.net' Method: OPTIONS [Sep 11 10:08:11] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/EiconSoftIP [Sep 11 10:08:11] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - EiconSoftIP [Sep 11 10:08:11] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer EiconSoftIP [Sep 11 10:08:11] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/EiconSoftIP - state 1 (Not in use) [Sep 11 10:08:11] DEBUG[18541]: app_queue.c:548 changethread: Device 'SIP/EiconSoftIP' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.3:5060 ---> REGISTER sip:pbx.example.net;transport=UDP SIP/2.0 From: ;tag=1397498-0-13c4-152cd0-72faf74f-152cd0 To: Call-ID: 13a2e28-0-13c4-152cd0-596ffaae-152cd0 CSeq: 26269 REGISTER Via: SIP/2.0/UDP 192.168.161.3:5060;branch=z9hG4bK-2ca42c-ae614bfe-2cf3e9a6 Max-Forwards: 70 Supported: replaces,timer,100rel Contact: Expires: 120 Authorization: Digest username="acMP404",realm="siprealm",nonce="0890d66b",uri="sip:pbx.example.net;transport=UDP",response="5393ad8db9d7daf97019c95a8ca14fbf",algorithm=MD5 Content-Length: 0 <-------------> [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net;transport=UDP SIP/2.0 (50) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: From: ;tag=1397498-0-13c4-152cd0-72faf74f-152cd0 (77) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: To: (33) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: Call-ID: 13a2e28-0-13c4-152cd0-596ffaae-152cd0 (46) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: CSeq: 26269 REGISTER (20) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Via: SIP/2.0/UDP 192.168.161.3:5060;branch=z9hG4bK-2ca42c-ae614bfe-2cf3e9a6 (75) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Supported: replaces,timer,100rel (32) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Contact: (41) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Expires: 120 (12) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Authorization: Digest username="acMP404",realm="siprealm",nonce="0890d66b",uri="sip:pbx.example.net;transport=UDP",response="5393ad8db9d7daf97019c95a8ca14fbf",algorithm=MD5 (172) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Length: 0 (17) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for 13a2e28-0-13c4-152cd0-596ffaae-152cd0 - REGISTER (No RTP) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.3 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.161.3:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.3:5060;branch=z9hG4bK-2ca42c-ae614bfe-2cf3e9a6;received=192.168.161.3 From: ;tag=1397498-0-13c4-152cd0-72faf74f-152cd0 To: Call-ID: 13a2e28-0-13c4-152cd0-596ffaae-152cd0 CSeq: 26269 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.161.3:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.161.3:5060;branch=z9hG4bK-2ca42c-ae614bfe-2cf3e9a6;received=192.168.161.3 From: ;tag=1397498-0-13c4-152cd0-72faf74f-152cd0 To: ;tag=as280aa775 Call-ID: 13a2e28-0-13c4-152cd0-596ffaae-152cd0 CSeq: 26269 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="siprealm", nonce="02c5b3dc" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '13a2e28-0-13c4-152cd0-596ffaae-152cd0' in 32000 ms (Method: REGISTER) <--- SIP read from 192.168.161.3:5060 ---> REGISTER sip:pbx.example.net;transport=UDP SIP/2.0 From: ;tag=1397498-0-13c4-152cd0-72faf74f-152cd0 To: Call-ID: 13a2e28-0-13c4-152cd0-596ffaae-152cd0 CSeq: 26270 REGISTER Via: SIP/2.0/UDP 192.168.161.3:5060;branch=z9hG4bK-2ca42c-ae614c26-11d90ca7 Max-Forwards: 70 Supported: replaces,timer,100rel Contact: Expires: 120 Authorization: Digest username="acMP404",realm="siprealm",nonce="02c5b3dc",uri="sip:pbx.example.net;transport=UDP",response="d5f9e1bc5b83bd25795e68462eaacd81",algorithm=MD5 Content-Length: 0 <-------------> [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net;transport=UDP SIP/2.0 (50) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: From: ;tag=1397498-0-13c4-152cd0-72faf74f-152cd0 (77) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: To: (33) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: Call-ID: 13a2e28-0-13c4-152cd0-596ffaae-152cd0 (46) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: CSeq: 26270 REGISTER (20) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Via: SIP/2.0/UDP 192.168.161.3:5060;branch=z9hG4bK-2ca42c-ae614c26-11d90ca7 (75) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Supported: replaces,timer,100rel (32) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Contact: (41) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Expires: 120 (12) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Authorization: Digest username="acMP404",realm="siprealm",nonce="02c5b3dc",uri="sip:pbx.example.net;transport=UDP",response="d5f9e1bc5b83bd25795e68462eaacd81",algorithm=MD5 (172) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Length: 0 (17) [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Sep 11 10:08:14] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.3 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.161.3:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.3:5060;branch=z9hG4bK-2ca42c-ae614c26-11d90ca7;received=192.168.161.3 From: ;tag=1397498-0-13c4-152cd0-72faf74f-152cd0 To: Call-ID: 13a2e28-0-13c4-152cd0-596ffaae-152cd0 CSeq: 26270 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.161.3:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.3:5060;branch=z9hG4bK-2ca42c-ae614c26-11d90ca7;received=192.168.161.3 From: ;tag=1397498-0-13c4-152cd0-72faf74f-152cd0 To: ;tag=as280aa775 Call-ID: 13a2e28-0-13c4-152cd0-596ffaae-152cd0 CSeq: 26270 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 120 Contact: ;expires=120 Date: Tue, 11 Sep 2007 08:08:14 GMT Content-Length: 0 <------------> [Sep 11 10:08:14] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/acMP404 Scheduling destruction of SIP dialog '13a2e28-0-13c4-152cd0-596ffaae-152cd0' in 32000 ms (Method: REGISTER) [Sep 11 10:08:14] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - acMP404 [Sep 11 10:08:14] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer acMP404 [Sep 11 10:08:14] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/acMP404 - state 1 (Not in use) [Sep 11 10:08:14] DEBUG[18542]: app_queue.c:548 changethread: Device 'SIP/acMP404' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.250:5060 ---> BYE sip:701@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-3k3p8qexf10s;rport From: "Theo Belder" ;tag=bjxxjlf0hu To: ;tag=as449bf3fa Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 3 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom320/6.5.10 RTP-RxStat: Total_Rx_Pkts=1298,Rx_Pkts=1298,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=1333,Tx_Pkts=1333,Remote_Tx_Pkts=0 Content-Length: 0 <-------------> [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: BYE sip:701@192.168.161.100 SIP/2.0 (35) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-3k3p8qexf10s;rport (71) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Theo Belder" ;tag=bjxxjlf0hu (60) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=as449bf3fa (55) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC (55) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 3 BYE (11) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1 (63) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: User-Agent: snom320/6.5.10 (26) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: RTP-RxStat: Total_Rx_Pkts=1298,Rx_Pkts=1298,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 (80) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: RTP-TxStat: Total_Tx_Pkts=1333,Tx_Pkts=1333,Remote_Tx_Pkts=0 (60) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Length: 0 (17) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 192.168.161.250 : 5060 (NAT) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:1641 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 3c26823294ed-t2wihiydodry@snom320-0004132496DC [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:14407 handle_request_bye: Received bye, issuing owner hangup <--- Transmitting (NAT) to 192.168.161.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-3k3p8qexf10s;received=192.168.161.250;rport=5060 From: "Theo Belder" ;tag=bjxxjlf0hu To: ;tag=as449bf3fa Call-ID: 3c26823294ed-t2wihiydodry@snom320-0004132496DC CSeq: 3 BYE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Sep 11 10:08:16] DEBUG[18518]: rtp.c:2918 bridge_native_loop: Oooh, got a hangup [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:17153 sip_set_rtp_peer: Sending reinvite on SIP '16fb24774565e40c06ce92b510a92129@pbx.example.net' - It's audio soon redirected to IP 192.168.161.100 [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:5713 reqprep: Strict routing enforced for session 16fb24774565e40c06ce92b510a92129@pbx.example.net set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.247, port 5060 [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:6263 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:6264 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 192.168.161.100 port 19310 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:6395 add_sdp: -- Done with adding codecs to SDP [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:6440 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:1629 initialize_initreq: Initializing already initialized SIP dialog 16fb24774565e40c06ce92b510a92129@pbx.example.net (presumably reinvite) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 0: INVITE sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 (57) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK0879906b;rport (66) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=as456f0c98 (55) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=viejglm0sj (63) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 4: Contact: (34) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net (57) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 6: CSeq: 106 INVITE (16) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 9: Remote-Party-ID: "Rámon" ;privacy=off;screen=no (73) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 12: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 13: Content-Type: application/sdp (29) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 14: Content-Length: 246 (19) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4640 parse_request: Header 15: (0) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: o=root 18440 18444 IN IP4 192.168.161.100 (41) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: s=session (9) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.100 (24) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: m=audio 19310 RTP/AVP 8 101 (27) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=silenceSupp:off - - - - (25) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.161.247:5060: INVITE sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK0879906b;rport From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Contact: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 106 INVITE User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Rámon" ;privacy=off;screen=no Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 246 v=0 o=root 18440 18444 IN IP4 192.168.161.100 s=session c=IN IP4 192.168.161.100 t=0 0 m=audio 19310 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #256 [Sep 11 10:08:16] DEBUG[18518]: channel.c:4268 ast_channel_bridge: Returning from native bridge, channels: SIP/403-09334c20, SIP/404-0935a2b8 [Sep 11 10:08:16] DEBUG[18518]: channel.c:1764 ast_hangup: Hanging up channel 'SIP/404-0935a2b8' [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:3368 sip_hangup: Hangup call SIP/404-0935a2b8, SIP callid 16fb24774565e40c06ce92b510a92129@pbx.example.net) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:3377 sip_hangup: update_call_counter(404) - decrement call limit counter on hangup [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:3058 update_call_counter: Updating call counter for outgoing call [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:3107 update_call_counter: Call to peer '404' removed from call limit 30 [Sep 11 10:08:16] DEBUG[18518]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/404 [Sep 11 10:08:16] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:08:16] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:08:16] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/404 - state 1 (Not in use) [Sep 11 10:08:16] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:08:16] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 Scheduling destruction of SIP dialog '16fb24774565e40c06ce92b510a92129@pbx.example.net' in 32000 ms (Method: INVITE) [Sep 11 10:08:16] DEBUG[18518]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/404-0935a2b8 [Sep 11 10:08:16] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:08:16] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:08:16] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/404 - state 1 (Not in use) [Sep 11 10:08:16] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:08:16] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:08:16] DEBUG[18518]: cdr.c:1164 ast_cdr_detach: Dropping CDR ! [Sep 11 10:08:16] DEBUG[18518]: rtp.c:1499 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Sep 11 10:08:16] DEBUG[18518]: app_dial.c:1686 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Sep 11 10:08:16] DEBUG[18518]: pbx.c:2407 __ast_pbx_run: Spawn extension (user-01,704,1) exited non-zero on 'SIP/403-09334c20'  == Spawn extension (user-01, 704, 1) exited non-zero on 'SIP/403-09334c20' [Sep 11 10:08:16] DEBUG[18518]: channel.c:1547 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/403-09334c20' [Sep 11 10:08:16] DEBUG[18518]: channel.c:1764 ast_hangup: Hanging up channel 'SIP/403-09334c20' [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:3368 sip_hangup: Hangup call SIP/403-09334c20, SIP callid 3c26823294ed-t2wihiydodry@snom320-0004132496DC) [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:3377 sip_hangup: update_call_counter(403) - decrement call limit counter on hangup [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:3058 update_call_counter: Updating call counter for incoming call [Sep 11 10:08:16] DEBUG[18518]: chan_sip.c:3107 update_call_counter: Call from peer '403' removed from call limit 30 [Sep 11 10:08:16] DEBUG[18518]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/403 [Sep 11 10:08:16] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:08:16] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:08:16] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/403 - state 1 (Not in use) [Sep 11 10:08:16] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:08:16] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:08:16] DEBUG[18518]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/403-09334c20 [Sep 11 10:08:16] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:08:16] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:08:16] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/403 - state 1 (Not in use) [Sep 11 10:08:16] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:08:16] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:08:16] DEBUG[18543]: app_queue.c:548 changethread: Device 'SIP/404' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 11 10:08:16] DEBUG[18544]: app_queue.c:548 changethread: Device 'SIP/404' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 11 10:08:16] DEBUG[18545]: app_queue.c:548 changethread: Device 'SIP/403' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 11 10:08:16] DEBUG[18546]: app_queue.c:548 changethread: Device 'SIP/403' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.247:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK0879906b;rport=5060 From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 106 INVITE Contact: ;flow-id=1 User-Agent: snom360/6.5.10 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 224 v=0 o=root 500070299 500070304 IN IP4 192.168.161.247 s=call c=IN IP4 192.168.161.247 t=0 0 m=audio 18596 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 Ok (14) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK0879906b;rport=5060 (71) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=as456f0c98 (55) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=viejglm0sj (63) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net (57) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 106 INVITE (16) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom360/6.5.10 (26) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Allow-Events: talk, hold, refer (31) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Supported: timer, 100rel, replaces, callerid (44) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Content-Type: application/sdp (29) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 224 (19) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: (0) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: v=0 (3) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: o=root 500070299 500070304 IN IP4 192.168.161.247 (49) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: s=call (6) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: c=IN IP4 192.168.161.247 (24) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: t=0 0 (5) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: m=audio 18596 RTP/AVP 8 101 (27) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:8 pcma/8000 (20) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=fmtp:101 0-16 (15) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=ptime:20 (10) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4672 parse_request: Line: a=sendrecv (10) --- (13 headers 11 lines) --- [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:2120 __sip_ack: Acked pending invite 106 [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #256 [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '16fb24774565e40c06ce92b510a92129@pbx.example.net' of Request 106: Match Not Found [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:11739 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.161.247:18596 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:5226 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.161.247:18596 [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:5306 process_sdp: We're settling with these formats: 0x8 (alaw) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:3058 update_call_counter: Updating call counter for outgoing call [Sep 11 10:08:16] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/404 [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:8004 build_route: build_route: Retaining previous route: [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 16fb24774565e40c06ce92b510a92129@pbx.example.net set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.247, port 5060 Transmitting (no NAT) to 192.168.161.247:5060: ACK sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK7b760998;rport From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Contact: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 106 ACK User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Rámon" ;privacy=off;screen=no Content-Length: 0 --- [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:5713 reqprep: Strict routing enforced for session 16fb24774565e40c06ce92b510a92129@pbx.example.net set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.161.247, port 5060 Reliably Transmitting (no NAT) to 192.168.161.247:5060: BYE sip:404@192.168.161.247:5060;line=llert2fa SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK6f36cae4;rport From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 107 BYE User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Rámon" ;privacy=off;screen=no Content-Length: 0 --- [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #258 Scheduling destruction of SIP dialog '16fb24774565e40c06ce92b510a92129@pbx.example.net' in 32000 ms (Method: INVITE) Really destroying SIP dialog '3c26823294ed-t2wihiydodry@snom320-0004132496DC' Method: BYE [Sep 11 10:08:16] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:08:16] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:08:16] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/404 - state 1 (Not in use) [Sep 11 10:08:16] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:08:16] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:08:16] DEBUG[18547]: app_queue.c:548 changethread: Device 'SIP/404' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.247:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK6f36cae4;rport=5060 From: "Rámon" ;tag=as456f0c98 To: ;tag=viejglm0sj Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net CSeq: 107 BYE Contact: ;flow-id=1 User-Agent: snom360/6.5.10 RTP-RxStat: Total_Rx_Pkts=751,Rx_Pkts=751,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=824,Tx_Pkts=824,Remote_Tx_Pkts=0 Content-Length: 0 <-------------> [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: SIP/2.0 200 OK (14) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK6f36cae4;rport=5060 (71) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=as456f0c98 (55) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: ;tag=viejglm0sj (63) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 16fb24774565e40c06ce92b510a92129@pbx.example.net (57) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 107 BYE (13) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Contact: ;flow-id=1 (63) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: snom360/6.5.10 (26) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: RTP-RxStat: Total_Rx_Pkts=751,Rx_Pkts=751,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 (78) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: RTP-TxStat: Total_Tx_Pkts=824,Tx_Pkts=824,Remote_Tx_Pkts=0 (58) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Content-Length: 0 (17) [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: (0) --- (11 headers 0 lines) --- [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:2128 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #258 [Sep 11 10:08:16] DEBUG[18453]: chan_sip.c:2138 __sip_ack: Stopping retransmission on '16fb24774565e40c06ce92b510a92129@pbx.example.net' of Request 107: Match Not Found Really destroying SIP dialog '16fb24774565e40c06ce92b510a92129@pbx.example.net' Method: INVITE <--- SIP read from 192.168.161.247:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-qmsbrd8tcp5s;rport From: "Sebastiaan" ;tag=00zbytvuew To: "Sebastiaan" Call-ID: 3c26701d900b-345tf3vzrgf8@snom360-000413231BA9 CSeq: 34 REGISTER Max-Forwards: 70 Contact: ;flow-id=1;q=1.0;+sip.instance="" User-Agent: snom360/6.5.10 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.161.247 WWW-Contact: WWW-Contact: Expires: 600 Content-Length: 0 <-------------> [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-qmsbrd8tcp5s;rport (71) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Sebastiaan" ;tag=00zbytvuew (59) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Sebastiaan" (42) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c26701d900b-345tf3vzrgf8@snom360-000413231BA9 (55) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 34 REGISTER (17) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1;q=1.0;+sip.instance="" (133) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: User-Agent: snom360/6.5.10 (26) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Supported: gruu (15) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow-Events: dialog (20) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: X-Real-IP: 192.168.161.247 (26) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: WWW-Contact: (40) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: WWW-Contact: (42) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: Expires: 600 (12) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 15: Content-Length: 0 (17) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 16: (0) --- (16 headers 0 lines) --- [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for 3c26701d900b-345tf3vzrgf8@snom360-000413231BA9 - REGISTER (No RTP) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.247 : 5060 (NAT) <--- Transmitting (no NAT) to 192.168.161.247:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-qmsbrd8tcp5s;received=192.168.161.247;rport=5060 From: "Sebastiaan" ;tag=00zbytvuew To: "Sebastiaan" Call-ID: 3c26701d900b-345tf3vzrgf8@snom360-000413231BA9 CSeq: 34 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.161.247:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-qmsbrd8tcp5s;received=192.168.161.247;rport=5060 From: "Sebastiaan" ;tag=00zbytvuew To: "Sebastiaan" ;tag=as1ae014e8 Call-ID: 3c26701d900b-345tf3vzrgf8@snom360-000413231BA9 CSeq: 34 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="siprealm", nonce="44825758" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c26701d900b-345tf3vzrgf8@snom360-000413231BA9' in 32000 ms (Method: REGISTER) <--- SIP read from 192.168.161.247:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-nvvp6ggssr45;rport From: "Sebastiaan" ;tag=00zbytvuew To: "Sebastiaan" Call-ID: 3c26701d900b-345tf3vzrgf8@snom360-000413231BA9 CSeq: 35 REGISTER Max-Forwards: 70 Contact: ;flow-id=1;q=1.0;+sip.instance="" User-Agent: snom360/6.5.10 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.161.247 WWW-Contact: WWW-Contact: Authorization: Digest username="404",realm="siprealm",nonce="44825758",uri="sip:pbx.example.net",response="18a956b79a6672cf5da7d796a88052c9",algorithm=MD5 Expires: 600 Content-Length: 0 <-------------> [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-nvvp6ggssr45;rport (71) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Sebastiaan" ;tag=00zbytvuew (59) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Sebastiaan" (42) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c26701d900b-345tf3vzrgf8@snom360-000413231BA9 (55) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 35 REGISTER (17) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1;q=1.0;+sip.instance="" (133) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: User-Agent: snom360/6.5.10 (26) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Supported: gruu (15) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow-Events: dialog (20) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: X-Real-IP: 192.168.161.247 (26) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: WWW-Contact: (40) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: WWW-Contact: (42) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: Authorization: Digest username="404",realm="siprealm",nonce="44825758",uri="sip:pbx.example.net",response="18a956b79a6672cf5da7d796a88052c9",algorithm=MD5 (154) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 15: Expires: 600 (12) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 16: Content-Length: 0 (17) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 17: (0) --- (17 headers 0 lines) --- [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.247 : 5060 (NAT) <--- Transmitting (no NAT) to 192.168.161.247:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-nvvp6ggssr45;received=192.168.161.247;rport=5060 From: "Sebastiaan" ;tag=00zbytvuew To: "Sebastiaan" Call-ID: 3c26701d900b-345tf3vzrgf8@snom360-000413231BA9 CSeq: 35 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------>  -- Saved useragent "snom360/6.5.10" for peer 404 <--- Transmitting (no NAT) to 192.168.161.247:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-nvvp6ggssr45;received=192.168.161.247;rport=5060 From: "Sebastiaan" ;tag=00zbytvuew To: "Sebastiaan" ;tag=as1ae014e8 Call-ID: 3c26701d900b-345tf3vzrgf8@snom360-000413231BA9 CSeq: 35 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 600 Contact: ;expires=600 Date: Tue, 11 Sep 2007 08:08:21 GMT Content-Length: 0 <------------> [Sep 11 10:08:21] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/404 Scheduling destruction of SIP dialog '3c26701d900b-345tf3vzrgf8@snom360-000413231BA9' in 32000 ms (Method: REGISTER) [Sep 11 10:08:21] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:08:21] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:08:21] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/404 - state 1 (Not in use) [Sep 11 10:08:21] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 404 [Sep 11 10:08:21] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 404 [Sep 11 10:08:21] DEBUG[18548]: app_queue.c:548 changethread: Device 'SIP/404' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.161.250:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-2x78ov36crrz;rport From: "Theo Belder" ;tag=digor5x0uy To: "Theo Belder" Call-ID: 3c2670c27530-20oaqjwf03na@snom320-0004132496DC CSeq: 34 REGISTER Max-Forwards: 70 Contact: ;flow-id=1;q=1.0;+sip.instance="" User-Agent: snom320/6.5.10 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.161.250 WWW-Contact: WWW-Contact: Expires: 600 Content-Length: 0 <-------------> [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-2x78ov36crrz;rport (71) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Theo Belder" ;tag=digor5x0uy (60) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Theo Belder" (43) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c2670c27530-20oaqjwf03na@snom320-0004132496DC (55) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 34 REGISTER (17) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1;q=1.0;+sip.instance="" (133) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: User-Agent: snom320/6.5.10 (26) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Supported: gruu (15) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow-Events: dialog (20) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: X-Real-IP: 192.168.161.250 (26) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: WWW-Contact: (40) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: WWW-Contact: (42) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: Expires: 600 (12) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 15: Content-Length: 0 (17) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 16: (0) --- (16 headers 0 lines) --- [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for 3c2670c27530-20oaqjwf03na@snom320-0004132496DC - REGISTER (No RTP) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.250 : 5060 (NAT) <--- Transmitting (no NAT) to 192.168.161.250:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-2x78ov36crrz;received=192.168.161.250;rport=5060 From: "Theo Belder" ;tag=digor5x0uy To: "Theo Belder" Call-ID: 3c2670c27530-20oaqjwf03na@snom320-0004132496DC CSeq: 34 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.161.250:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-2x78ov36crrz;received=192.168.161.250;rport=5060 From: "Theo Belder" ;tag=digor5x0uy To: "Theo Belder" ;tag=as2ad7dadb Call-ID: 3c2670c27530-20oaqjwf03na@snom320-0004132496DC CSeq: 34 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="siprealm", nonce="53252291" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c2670c27530-20oaqjwf03na@snom320-0004132496DC' in 32000 ms (Method: REGISTER) <--- SIP read from 192.168.161.246:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-6bd23p7tyguj;rport From: "Rámon" ;tag=gjw2yz292z To: "Rámon" Call-ID: 3c26707aa604-sj8hp12lm5pu@snom360-000413231BC4 CSeq: 34 REGISTER Max-Forwards: 70 Contact: ;flow-id=1;q=1.0;+sip.instance="" User-Agent: snom360/6.5.10 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.161.246 WWW-Contact: WWW-Contact: Expires: 600 Content-Length: 0 <-------------> [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-6bd23p7tyguj;rport (71) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=gjw2yz292z (55) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Rámon" (38) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c26707aa604-sj8hp12lm5pu@snom360-000413231BC4 (55) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 34 REGISTER (17) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1;q=1.0;+sip.instance="" (133) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: User-Agent: snom360/6.5.10 (26) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Supported: gruu (15) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow-Events: dialog (20) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: X-Real-IP: 192.168.161.246 (26) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: WWW-Contact: (40) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: WWW-Contact: (42) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: Expires: 600 (12) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 15: Content-Length: 0 (17) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 16: (0) --- (16 headers 0 lines) --- [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for 3c26707aa604-sj8hp12lm5pu@snom360-000413231BC4 - REGISTER (No RTP) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.246 : 5060 (NAT) <--- Transmitting (no NAT) to 192.168.161.246:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-6bd23p7tyguj;received=192.168.161.246;rport=5060 From: "Rámon" ;tag=gjw2yz292z To: "Rámon" Call-ID: 3c26707aa604-sj8hp12lm5pu@snom360-000413231BC4 CSeq: 34 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.161.246:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-6bd23p7tyguj;received=192.168.161.246;rport=5060 From: "Rámon" ;tag=gjw2yz292z To: "Rámon" ;tag=as54632def Call-ID: 3c26707aa604-sj8hp12lm5pu@snom360-000413231BC4 CSeq: 34 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="siprealm", nonce="382c7e9f" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c26707aa604-sj8hp12lm5pu@snom360-000413231BC4' in 32000 ms (Method: REGISTER) <--- SIP read from 192.168.161.250:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-6i53gf5x6xlm;rport From: "Theo Belder" ;tag=digor5x0uy To: "Theo Belder" Call-ID: 3c2670c27530-20oaqjwf03na@snom320-0004132496DC CSeq: 35 REGISTER Max-Forwards: 70 Contact: ;flow-id=1;q=1.0;+sip.instance="" User-Agent: snom320/6.5.10 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.161.250 WWW-Contact: WWW-Contact: Authorization: Digest username="403",realm="siprealm",nonce="53252291",uri="sip:pbx.example.net",response="17b4c2b1cc3d4318158565d4b028a0b8",algorithm=MD5 Expires: 600 Content-Length: 0 <-------------> [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-6i53gf5x6xlm;rport (71) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Theo Belder" ;tag=digor5x0uy (60) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Theo Belder" (43) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c2670c27530-20oaqjwf03na@snom320-0004132496DC (55) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 35 REGISTER (17) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1;q=1.0;+sip.instance="" (133) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: User-Agent: snom320/6.5.10 (26) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Supported: gruu (15) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow-Events: dialog (20) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: X-Real-IP: 192.168.161.250 (26) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: WWW-Contact: (40) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: WWW-Contact: (42) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: Authorization: Digest username="403",realm="siprealm",nonce="53252291",uri="sip:pbx.example.net",response="17b4c2b1cc3d4318158565d4b028a0b8",algorithm=MD5 (154) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 15: Expires: 600 (12) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 16: Content-Length: 0 (17) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 17: (0) --- (17 headers 0 lines) --- [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.250 : 5060 (NAT) <--- Transmitting (no NAT) to 192.168.161.250:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-6i53gf5x6xlm;received=192.168.161.250;rport=5060 From: "Theo Belder" ;tag=digor5x0uy To: "Theo Belder" Call-ID: 3c2670c27530-20oaqjwf03na@snom320-0004132496DC CSeq: 35 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------>  -- Saved useragent "snom320/6.5.10" for peer 403 <--- Transmitting (no NAT) to 192.168.161.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-6i53gf5x6xlm;received=192.168.161.250;rport=5060 From: "Theo Belder" ;tag=digor5x0uy To: "Theo Belder" ;tag=as2ad7dadb Call-ID: 3c2670c27530-20oaqjwf03na@snom320-0004132496DC CSeq: 35 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 600 Contact: ;expires=600 Date: Tue, 11 Sep 2007 08:08:21 GMT Content-Length: 0 <------------> [Sep 11 10:08:21] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/403 Scheduling destruction of SIP dialog '3c2670c27530-20oaqjwf03na@snom320-0004132496DC' in 32000 ms (Method: REGISTER) [Sep 11 10:08:21] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:08:21] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:08:21] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/403 - state 1 (Not in use) [Sep 11 10:08:21] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 403 [Sep 11 10:08:21] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 403 [Sep 11 10:08:21] DEBUG[18549]: app_queue.c:548 changethread: Device 'SIP/403' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: OPTIONS sip:192.168.1.160 SIP/2.0 (33) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4861ab7a;rport (66) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "asterisk" ;tag=as340f1953 (62) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: (23) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Contact: (39) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: Call-ID: 1a8776990b6b6535568fb98c48575494@pbx.example.net (57) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: User-Agent: atCOM PBX (21) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Date: Tue, 11 Sep 2007 08:08:21 GMT (35) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: Supported: replaces (19) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4861ab7a;rport From: "asterisk" ;tag=as340f1953 To: Contact: Call-ID: 1a8776990b6b6535568fb98c48575494@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:2017 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #267 <--- SIP read from 192.168.161.246:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-3o9kr4a2id5r;rport From: "Rámon" ;tag=gjw2yz292z To: "Rámon" Call-ID: 3c26707aa604-sj8hp12lm5pu@snom360-000413231BC4 CSeq: 35 REGISTER Max-Forwards: 70 Contact: ;flow-id=1;q=1.0;+sip.instance="" User-Agent: snom360/6.5.10 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.161.246 WWW-Contact: WWW-Contact: Authorization: Digest username="401",realm="siprealm",nonce="382c7e9f",uri="sip:pbx.example.net",response="2b51f15973606d9662168afbc528235e",algorithm=MD5 Expires: 600 Content-Length: 0 <-------------> [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-3o9kr4a2id5r;rport (71) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 2: From: "Rámon" ;tag=gjw2yz292z (55) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 3: To: "Rámon" (38) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 4: Call-ID: 3c26707aa604-sj8hp12lm5pu@snom360-000413231BC4 (55) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 5: CSeq: 35 REGISTER (17) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 7: Contact: ;flow-id=1;q=1.0;+sip.instance="" (133) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 8: User-Agent: snom360/6.5.10 (26) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 9: Supported: gruu (15) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 10: Allow-Events: dialog (20) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 11: X-Real-IP: 192.168.161.246 (26) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 12: WWW-Contact: (40) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 13: WWW-Contact: (42) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 14: Authorization: Digest username="401",realm="siprealm",nonce="382c7e9f",uri="sip:pbx.example.net",response="2b51f15973606d9662168afbc528235e",algorithm=MD5 (154) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 15: Expires: 600 (12) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 16: Content-Length: 0 (17) [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:4640 parse_request: Header 17: (0) --- (17 headers 0 lines) --- [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:14865 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.161.246 : 5060 (NAT) <--- Transmitting (no NAT) to 192.168.161.246:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-3o9kr4a2id5r;received=192.168.161.246;rport=5060 From: "Rámon" ;tag=gjw2yz292z To: "Rámon" Call-ID: 3c26707aa604-sj8hp12lm5pu@snom360-000413231BC4 CSeq: 35 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------>  -- Saved useragent "snom360/6.5.10" for peer 401 <--- Transmitting (no NAT) to 192.168.161.246:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.246:5060;branch=z9hG4bK-3o9kr4a2id5r;received=192.168.161.246;rport=5060 From: "Rámon" ;tag=gjw2yz292z To: "Rámon" ;tag=as54632def Call-ID: 3c26707aa604-sj8hp12lm5pu@snom360-000413231BC4 CSeq: 35 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 600 Contact: ;expires=600 Date: Tue, 11 Sep 2007 08:08:21 GMT Content-Length: 0 <------------> [Sep 11 10:08:21] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/401 [Sep 11 10:08:21] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:08:21] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 [Sep 11 10:08:21] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/401 - state 1 (Not in use) [Sep 11 10:08:21] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 401 [Sep 11 10:08:21] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer 401 Scheduling destruction of SIP dialog '3c26707aa604-sj8hp12lm5pu@snom360-000413231BC4' in 32000 ms (Method: REGISTER) [Sep 11 10:08:21] DEBUG[18550]: app_queue.c:548 changethread: Device 'SIP/401' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:2059 __sip_autodestruct: Auto destroying SIP dialog '9713438371120002023@192.168.161.2' [Sep 11 10:08:21] DEBUG[18453]: chan_sip.c:3165 sip_destroy: Destroying SIP dialog 9713438371120002023@192.168.161.2 Really destroying SIP dialog '9713438371120002023@192.168.161.2' Method: REGISTER [Sep 11 10:08:22] DEBUG[18453]: chan_sip.c:1879 retrans_pkt: SIP TIMER: Not rescheduling id #267:OPTIONS (Method 3) (No timer T1) Retransmitting #1 (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4861ab7a;rport From: "asterisk" ;tag=as340f1953 To: Contact: Call-ID: 1a8776990b6b6535568fb98c48575494@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:23] DEBUG[18453]: chan_sip.c:1879 retrans_pkt: SIP TIMER: Not rescheduling id #267:OPTIONS (Method 3) (No timer T1) Retransmitting #2 (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4861ab7a;rport From: "asterisk" ;tag=as340f1953 To: Contact: Call-ID: 1a8776990b6b6535568fb98c48575494@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- s[Sep 11 10:08:24] DEBUG[18453]: chan_sip.c:1879 retrans_pkt: SIP TIMER: Not rescheduling id #267:OPTIONS (Method 3) (No timer T1) Retransmitting #3 (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4861ab7a;rport From: "asterisk" ;tag=as340f1953 To: Contact: Call-ID: 1a8776990b6b6535568fb98c48575494@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- top [Sep 11 10:08:25] DEBUG[18453]: chan_sip.c:1879 retrans_pkt: SIP TIMER: Not rescheduling id #267:OPTIONS (Method 3) (No timer T1) Retransmitting #4 (no NAT) to 192.168.1.160:5060: OPTIONS sip:192.168.1.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4861ab7a;rport From: "asterisk" ;tag=as340f1953 To: Contact: Call-ID: 1a8776990b6b6535568fb98c48575494@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Tue, 11 Sep 2007 08:08:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Sep 11 10:08:25] DEBUG[18453]: chan_sip.c:3165 sip_destroy: Destroying SIP dialog 1a8776990b6b6535568fb98c48575494@pbx.example.net Really destroying SIP dialog '1a8776990b6b6535568fb98c48575494@pbx.example.net' Method: OPTIONS [Sep 11 10:08:25] DEBUG[18453]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/EiconSoftIP [Sep 11 10:08:25] DEBUG[18445]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - EiconSoftIP [Sep 11 10:08:25] DEBUG[18445]: chan_sip.c:15490 sip_devicestate: Checking device state for peer EiconSoftIP [Sep 11 10:08:25] DEBUG[18445]: devicestate.c:287 do_state_change: Changing state for SIP/EiconSoftIP - state 1 (Not in use) [Sep 11 10:08:25] DEBUG[18551]: app_queue.c:548 changethread: Device 'SIP/EiconSoftIP' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. now Beginning asterisk shutdown.... Executing last minute cleanups  == Destroying musiconhold processes Asterisk cleanly ending (0). [Sep 11 10:08:26] DEBUG[18440]: asterisk.c:1278 quit_handler: Asterisk ending (0).