[Oct 22 16:36:14] VERBOSE[1874] config.c: == Parsing '/etc/asterisk/logger.conf': [Oct 22 16:36:14] DEBUG[1874] config.c: Parsing /etc/asterisk/logger.conf [Oct 22 16:36:14] VERBOSE[1874] config.c: == Found [Oct 22 16:36:14] VERBOSE[1874] logger.c: Asterisk Queue Logger restarted [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.203:2051 ---> INVITE sip:150@192.168.10.70;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.203:2051;branch=z9hG4bK-8d2yjmrqfsms;rport From: "Erika" ;tag=3vqvxksyq8 To: Call-ID: 3c26c3d24efb-50v3t7e4o9ys CSeq: 1 INVITE Max-Forwards: 70 Contact: ;reg-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/7.3.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 330 v=0 o=root 888790008 888790008 IN IP4 192.168.10.203 s=call c=IN IP4 192.168.10.203 t=0 0 m=audio 56056 RTP/AVP 8 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:k0VIDWJVRkXgWSOScrIcMTu1KbwpzEQq6+4apxBo a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 0 [ 47]: INVITE sip:150@192.168.10.70;user=phone SIP/2.0 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.10.203:2051;branch=z9hG4bK-8d2yjmrqfsms;rport [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 2 [ 55]: From: "Erika" ;tag=3vqvxksyq8 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 3 [ 38]: To: [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 4 [ 34]: Call-ID: 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 7 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 8 [ 41]: P-Key-Flags: resolution="31x13", keys="4" [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 9 [ 26]: User-Agent: snom360/7.3.30 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 11 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 12 [ 42]: Allow-Events: talk, hold, refer, call-info [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 13 [ 47]: Supported: timer, 100rel, replaces, from-change [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 14 [ 35]: Session-Expires: 3600;refresher=uas [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 15 [ 10]: Min-SE: 90 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 16 [ 29]: Content-Type: application/sdp [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 17 [ 19]: Content-Length: 330 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 18 [ 0]: [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Body 0 [ 3]: v=0 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Body 1 [ 48]: o=root 888790008 888790008 IN IP4 192.168.10.203 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Body 2 [ 6]: s=call [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.203 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Body 5 [ 29]: m=audio 56056 RTP/AVP 8 0 101 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:k0VIDWJVRkXgWSOScrIcMTu1KbwpzEQq6+4apxBo [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 pcma/8000 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 pcmu/8000 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Body 12 [ 10]: a=sendrecv [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: --- (18 headers 13 lines) --- [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: = Looking for Call ID: 3c26c3d24efb-50v3t7e4o9ys (Checking From) --From tag 3vqvxksyq8 --To-tag [Oct 22 16:37:01] DEBUG[1852] acl.c: For destination '192.168.10.203', our source address is '192.168.10.70'. [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Allocating new SIP dialog for 3c26c3d24efb-50v3t7e4o9ys - INVITE (No RTP) [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Oct 22 16:37:01] DEBUG[1852] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change" [Oct 22 16:37:01] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -timer- [Oct 22 16:37:01] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: timer [Oct 22 16:37:01] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -100rel- [Oct 22 16:37:01] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: 100rel [Oct 22 16:37:01] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -replaces- [Oct 22 16:37:01] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: replaces [Oct 22 16:37:01] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -from-change- [Oct 22 16:37:01] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: from-change [Oct 22 16:37:01] DEBUG[1852] netsock2.c: Splitting '192.168.10.203:2051' gives... [Oct 22 16:37:01] DEBUG[1852] netsock2.c: ...host '192.168.10.203' and port '2051'. [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: Sending to 192.168.10.203:2051 (no NAT) [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Initializing initreq for method INVITE - callid 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: Using INVITE request as basis request - 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: Found peer 'phone1' for 'phone1' from 192.168.10.203:2051 [Oct 22 16:37:01] DEBUG[1852] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xcd270a0' [Oct 22 16:37:01] DEBUG[1852] res_rtp_asterisk.c: Allocated port 19452 for RTP instance '0xcd270a0' [Oct 22 16:37:01] DEBUG[1852] rtp_engine.c: RTP instance '0xcd270a0' is setup and ready to go [Oct 22 16:37:01] DEBUG[1852] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xcd270a0' [Oct 22 16:37:01] VERBOSE[1852] netsock2.c: == Using SIP RTP TOS bits 184 [Oct 22 16:37:01] VERBOSE[1852] netsock2.c: == Using SIP RTP CoS mark 5 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Setting NAT on RTP to Off [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 888790008 888790008 IN IP4 192.168.10.203... UNSUPPORTED. [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 16:37:01] DEBUG[1852] netsock2.c: Splitting '192.168.10.203' gives... [Oct 22 16:37:01] DEBUG[1852] netsock2.c: ...host '192.168.10.203' and port '(null)'. [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: Found RTP audio format 8 [Oct 22 16:37:01] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33a9f88 [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: Found RTP audio format 0 [Oct 22 16:37:01] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33a9f88 [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: Found RTP audio format 101 [Oct 22 16:37:01] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33a9f88 [Oct 22 16:37:01] ERROR[1852] chan_sip.c: No SRTP module loaded, can't setup SRTP session. [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:k0VIDWJVRkXgWSOScrIcMTu1KbwpzEQq6+4apxBo... UNSUPPORTED. [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: Found audio description format pcma for ID 8 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: Found audio description format pcmu for ID 0 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 16:37:01] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33a9f88 [Oct 22 16:37:01] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33a9f88 [Oct 22 16:37:01] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33a9f88 [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 22 16:37:01] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd270a0' [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: Peer audio RTP is at port 192.168.10.203:56056 [Oct 22 16:37:01] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33a9f88 to 0xcd2724c [Oct 22 16:37:01] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33a9f88 to 0xcd2724c [Oct 22 16:37:01] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33a9f88 to 0xcd2724c [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Checking SIP call limits for device phone1 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Updating call counter for incoming call [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Call from peer 'phone1' is 1 out of 2147483647 [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: Looking for 150 in Standard (domain 192.168.10.70) [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: *** Our native formats are 0x8 (alaw) [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: This channel will not be able to handle video. [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: list_route: hop: [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Session-Expires: 3600 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Refresher: UAS [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Received Min-SE: 90 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Session timer started: 378 - 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: SIP/phone1-00000016: New call is still down.... Trying... [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.203:2051 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.203:2051;branch=z9hG4bK-8d2yjmrqfsms;received=192.168.10.203;rport=2051 From: "Erika" ;tag=3vqvxksyq8 To: Call-ID: 3c26c3d24efb-50v3t7e4o9ys CSeq: 1 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.203:2051;branch=z9hG4bK-8d2yjmrqfsms;received=192.168.10.203;rport=2051 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 2 [ 55]: From: "Erika" ;tag=3vqvxksyq8 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 3 [ 38]: To: [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 4 [ 34]: Call-ID: 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 10 [ 37]: Contact: [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 12 [ 0]: [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 16:37:01] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 16:37:01] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 16:37:01] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 16:37:01] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 16:37:01] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 16:37:01] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 16:37:01] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 16:37:01] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 16:37:01] DEBUG[1854] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone1-00000016 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: Erika Musterfrau AccountCode: Exten: 150 Context: Standard Uniqueid: 1287758221.24 [Oct 22 16:37:01] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone1-00000016 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1287758221.24 [Oct 22 16:37:01] DEBUG[2187] pbx.c: Launching 'Dial' [Oct 22 16:37:01] VERBOSE[2187] pbx.c: -- Executing [150@Standard:1] Dial("SIP/phone1-00000016", "SIP/phone2") in new stack [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Allocating new SIP dialog for 090c67a153832aba1c6aa0d2103440c8@192.168.10.70 - INVITE (No RTP) [Oct 22 16:37:01] DEBUG[2187] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xcc440d0' [Oct 22 16:37:01] DEBUG[2187] res_rtp_asterisk.c: Allocated port 14778 for RTP instance '0xcc440d0' [Oct 22 16:37:01] DEBUG[2187] rtp_engine.c: RTP instance '0xcc440d0' is setup and ready to go [Oct 22 16:37:01] DEBUG[2187] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xcc440d0' [Oct 22 16:37:01] VERBOSE[2187] netsock2.c: == Using SIP RTP TOS bits 184 [Oct 22 16:37:01] VERBOSE[2187] netsock2.c: == Using SIP RTP CoS mark 5 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Setting NAT on RTP to Off [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Oct 22 16:37:01] DEBUG[2187] acl.c: For destination '192.168.10.204', our source address is '192.168.10.70'. [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: *** Our native formats are 0x8 (alaw) [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: This channel will not be able to handle video. [Oct 22 16:37:01] DEBUG[2187] rtp_engine.c: Seeded SDP of 'SIP/phone2-00000017' with that of 'SIP/phone1-00000016' [Oct 22 16:37:01] DEBUG[2187] channel.c: Not copying variable DIALEDTIME. [Oct 22 16:37:01] DEBUG[2187] channel.c: Not copying variable ANSWEREDTIME. [Oct 22 16:37:01] DEBUG[2187] channel.c: Not copying variable DIALEDPEERNAME. [Oct 22 16:37:01] DEBUG[2187] channel.c: Not copying variable DIALEDPEERNUMBER. [Oct 22 16:37:01] DEBUG[2187] channel.c: Not copying variable DIALSTATUS. [Oct 22 16:37:01] DEBUG[2187] channel.c: Not copying variable SIPCALLID. [Oct 22 16:37:01] DEBUG[2187] channel.c: Not copying variable SIPDOMAIN. [Oct 22 16:37:01] DEBUG[2187] channel.c: Not copying variable SIPURI. [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Outgoing Call for phone2 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Updating call counter for outgoing call [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Call to peer 'phone2' is 1 out of 2147483647 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 16:37:01] VERBOSE[2187] chan_sip.c: Audio is at 5060 [Oct 22 16:37:01] VERBOSE[2187] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Oct 22 16:37:01] VERBOSE[2187] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Oct 22 16:37:01] VERBOSE[2187] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Initializing initreq for method INVITE - callid 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.204:2050;line=xguiqus6 SIP/2.0 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK09402f85 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 4 [ 50]: To: [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 5 [ 37]: Contact: [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 6 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 9 [ 35]: Date: Fri, 22 Oct 2010 14:37:01 GMT [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Oct 22 16:37:01] VERBOSE[2187] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.204:2050: INVITE sip:phone2@192.168.10.204:2050;line=xguiqus6 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK09402f85 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as3c058282 To: Contact: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.0-1 Date: Fri, 22 Oct 2010 14:37:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "Erika Musterfrau" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1862472553 1862472553 IN IP4 192.168.10.70 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.70 t=0 0 m=audio 14778 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.204:2050;line=xguiqus6 SIP/2.0 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK09402f85 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 4 [ 50]: To: [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 5 [ 37]: Contact: [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 6 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 9 [ 35]: Date: Fri, 22 Oct 2010 14:37:01 GMT [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 14 [ 19]: Content-Length: 262 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 15 [ 0]: [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Body 0 [ 3]: v=0 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Body 1 [ 49]: o=root 1862472553 1862472553 IN IP4 192.168.10.70 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Body 5 [ 29]: m=audio 14778 RTP/AVP 8 0 101 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Body 11 [ 10]: a=sendrecv [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #380 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 16:37:01] VERBOSE[2187] app_dial.c: -- Called phone2 [Oct 22 16:37:01] DEBUG[1854] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone2-00000017 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 150 CallerIDName: Hans Muster AccountCode: Exten: Context: Standard Uniqueid: 1287758221.25 [Oct 22 16:37:01] DEBUG[1854] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/phone1-00000016 Destination: SIP/phone2-00000017 CallerIDNum: 100 CallerIDName: Erika Musterfrau UniqueID: 1287758221.24 DestUniqueID: 1287758221.25 Dialstring: phone2 [Oct 22 16:37:01] DEBUG[1854] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone2-00000017 CallerIDNum: 150 CallerIDName: Uniqueid: 1287758221.25 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Oct 22 16:37:01] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 16:37:01] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 16:37:01] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Oct 22 16:37:01] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '6' [Oct 22 16:37:01] DEBUG[1846] app_queue.c: Extension '100@Standard' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 22 16:37:01] DEBUG[1872] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 22 16:37:01] DEBUG[1872] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 22 16:37:01] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: Standard Hint: SIP/phone1 Status: 1 [Oct 22 16:37:01] DEBUG[1846] app_queue.c: Extension '150@Standard' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Oct 22 16:37:01] DEBUG[1872] app_queue.c: Device 'SIP/phone2' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Oct 22 16:37:01] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 8 [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.204:2050 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK09402f85 From: "Erika Musterfrau" ;tag=as3c058282 To: ;tag=ryuq3pvs4i Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 102 INVITE Contact: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK09402f85 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 3 [ 65]: To: ;tag=ryuq3pvs4i [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 4 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 8 [ 42]: Allow-Events: talk, hold, refer, call-info [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Oct 22 16:37:01] VERBOSE[1852] chan_sip.c: --- (10 headers 0 lines) --- [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: = Looking for Call ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 (Checking To) --From tag as3c058282 --To-tag ryuq3pvs4i [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: *** SIP TIMER: Cancelling retransmission #380 - INVITE (got response) [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Request 102: Found [Oct 22 16:37:01] DEBUG[1852] chan_sip.c: SIP response 180 to standard invite [Oct 22 16:37:01] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 16:37:01] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 16:37:01] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Oct 22 16:37:01] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '6' [Oct 22 16:37:01] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-00000017 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 150 CallerIDName: Uniqueid: 1287758221.25 [Oct 22 16:37:01] VERBOSE[2187] app_dial.c: -- SIP/phone2-00000017 is ringing [Oct 22 16:37:01] DEBUG[2187] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-00000016' with that of 'SIP/phone2-00000017' [Oct 22 16:37:01] VERBOSE[2187] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.203:2051 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.203:2051;branch=z9hG4bK-8d2yjmrqfsms;received=192.168.10.203;rport=2051 From: "Erika" ;tag=3vqvxksyq8 To: ;tag=as388f7611 Call-ID: 3c26c3d24efb-50v3t7e4o9ys CSeq: 1 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.203:2051;branch=z9hG4bK-8d2yjmrqfsms;received=192.168.10.203;rport=2051 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 2 [ 55]: From: "Erika" ;tag=3vqvxksyq8 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 3 [ 53]: To: ;tag=as388f7611 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 4 [ 34]: Call-ID: 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 10 [ 37]: Contact: [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Header 12 [ 0]: [Oct 22 16:37:01] DEBUG[2187] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 16:37:01] DEBUG[1872] app_queue.c: Device 'SIP/phone2' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Oct 22 16:37:02] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.204:2050 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK09402f85 From: "Erika Musterfrau" ;tag=as3c058282 To: ;tag=ryuq3pvs4i Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 102 INVITE Contact: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Oct 22 16:37:02] DEBUG[1852] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Oct 22 16:37:02] DEBUG[1852] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK09402f85 [Oct 22 16:37:02] DEBUG[1852] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:02] DEBUG[1852] chan_sip.c: Header 3 [ 65]: To: ;tag=ryuq3pvs4i [Oct 22 16:37:02] DEBUG[1852] chan_sip.c: Header 4 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:02] DEBUG[1852] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Oct 22 16:37:02] DEBUG[1852] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:02] DEBUG[1852] chan_sip.c: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Oct 22 16:37:02] DEBUG[1852] chan_sip.c: Header 8 [ 42]: Allow-Events: talk, hold, refer, call-info [Oct 22 16:37:02] DEBUG[1852] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Oct 22 16:37:02] VERBOSE[1852] chan_sip.c: --- (10 headers 0 lines) --- [Oct 22 16:37:02] DEBUG[1852] chan_sip.c: = Looking for Call ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 (Checking To) --From tag as3c058282 --To-tag ryuq3pvs4i [Oct 22 16:37:02] DEBUG[1852] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Request 102: Found [Oct 22 16:37:02] DEBUG[1852] chan_sip.c: SIP response 180 to standard invite [Oct 22 16:37:02] VERBOSE[2187] app_dial.c: -- SIP/phone2-00000017 is ringing [Oct 22 16:37:02] DEBUG[2187] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-00000016' with that of 'SIP/phone2-00000017' [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.204:2050 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK09402f85 From: "Erika Musterfrau" ;tag=as3c058282 To: ;tag=ryuq3pvs4i Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 102 INVITE Contact: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK09402f85 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 3 [ 65]: To: ;tag=ryuq3pvs4i [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 4 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 8 [ 42]: Allow-Events: talk, hold, refer, call-info [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: --- (10 headers 0 lines) --- [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: = Looking for Call ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 (Checking To) --From tag as3c058282 --To-tag ryuq3pvs4i [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Request 102: Found [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: SIP response 180 to standard invite [Oct 22 16:37:03] VERBOSE[2187] app_dial.c: -- SIP/phone2-00000017 is ringing [Oct 22 16:37:03] DEBUG[2187] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-00000016' with that of 'SIP/phone2-00000017' [Oct 22 16:37:03] DEBUG[2187] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Oct 22 16:37:03] DEBUG[1854] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.204:52689 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 16777216 FractionLost: 0 PacketsLost: 1 HighestSequence: 0 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Oct 22 16:37:03] DEBUG[2187] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcc440d0' [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.204:2050 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK09402f85 From: "Erika Musterfrau" ;tag=as3c058282 To: ;tag=ryuq3pvs4i Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 102 INVITE Contact: ;reg-id=1 User-Agent: snom320/7.3.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 246 v=0 o=root 514665578 514665579 IN IP4 192.168.10.204 s=call c=IN IP4 192.168.10.204 t=0 0 m=audio 52688 RTP/AVP 8 0 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK09402f85 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 3 [ 65]: To: ;tag=ryuq3pvs4i [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 4 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 7 [ 26]: User-Agent: snom320/7.3.30 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 8 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 9 [ 42]: Allow-Events: talk, hold, refer, call-info [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 10 [ 47]: Supported: timer, 100rel, replaces, from-change [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 12 [ 19]: Content-Length: 246 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 13 [ 0]: [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 0 [ 3]: v=0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 1 [ 48]: o=root 514665578 514665579 IN IP4 192.168.10.204 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 2 [ 6]: s=call [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.204 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 5 [ 29]: m=audio 52688 RTP/AVP 8 0 101 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 pcma/8000 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 pcmu/8000 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 11 [ 10]: a=sendrecv [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: --- (13 headers 12 lines) --- [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: = Looking for Call ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 (Checking To) --From tag as3c058282 --To-tag ryuq3pvs4i [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Acked pending invite 102 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Stopping retransmission on '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' of Request 102: Match Found [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: SIP response 200 to standard invite [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 514665578 514665579 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 16:37:03] DEBUG[1852] netsock2.c: Splitting '192.168.10.204' gives... [Oct 22 16:37:03] DEBUG[1852] netsock2.c: ...host '192.168.10.204' and port '(null)'. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found RTP audio format 8 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found RTP audio format 0 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found RTP audio format 101 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found audio description format pcma for ID 8 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found audio description format pcmu for ID 0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 22 16:37:03] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcc440d0' [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Peer audio RTP is at port 192.168.10.204:52688 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcc4427c [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcc4427c [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcc4427c [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: list_route: hop: [Oct 22 16:37:03] DEBUG[1852] netsock2.c: Splitting '192.168.10.204:2050' gives... [Oct 22 16:37:03] DEBUG[1852] netsock2.c: ...host '192.168.10.204' and port '2050'. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Strict routing enforced for session 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 22 16:37:03] DEBUG[1852] netsock2.c: Splitting '192.168.10.204:2050' gives... [Oct 22 16:37:03] DEBUG[1852] netsock2.c: ...host '192.168.10.204' and port '2050'. [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: set_destination: set destination to 192.168.10.204:2050 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Transmitting (no NAT) to 192.168.10.204:2050: ACK sip:phone2@192.168.10.204:2050;line=xguiqus6 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK670ddcdb Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as3c058282 To: ;tag=ryuq3pvs4i Contact: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 0 [ 56]: ACK sip:phone2@192.168.10.204:2050;line=xguiqus6 SIP/2.0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK670ddcdb [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 4 [ 65]: To: ;tag=ryuq3pvs4i [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 5 [ 37]: Contact: [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 6 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 10 [ 0]: [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 16:37:03] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 16:37:03] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 16:37:03] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 16:37:03] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 16:37:03] VERBOSE[2187] app_dial.c: -- SIP/phone2-00000017 answered SIP/phone1-00000016 [Oct 22 16:37:03] DEBUG[2187] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-00000016' with that of 'SIP/phone2-00000017' [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: SIP answering channel: SIP/phone1-00000016 [Oct 22 16:37:03] DEBUG[2187] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Setting framing from config on incoming call [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Oct 22 16:37:03] VERBOSE[2187] chan_sip.c: Audio is at 5060 [Oct 22 16:37:03] VERBOSE[2187] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Oct 22 16:37:03] VERBOSE[2187] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Oct 22 16:37:03] VERBOSE[2187] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 16:37:03] VERBOSE[2187] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.203:2051 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.203:2051;branch=z9hG4bK-8d2yjmrqfsms;received=192.168.10.203;rport=2051 From: "Erika" ;tag=3vqvxksyq8 To: ;tag=as388f7611 Call-ID: 3c26c3d24efb-50v3t7e4o9ys CSeq: 1 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: P-Asserted-Identity: "Hans Muster" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1411874980 1411874980 IN IP4 192.168.10.70 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.70 t=0 0 m=audio 19452 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.203:2051;branch=z9hG4bK-8d2yjmrqfsms;received=192.168.10.203;rport=2051 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 2 [ 55]: From: "Erika" ;tag=3vqvxksyq8 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 3 [ 53]: To: ;tag=as388f7611 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 4 [ 34]: Call-ID: 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 10 [ 37]: Contact: [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 11 [ 58]: P-Asserted-Identity: "Hans Muster" [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 13 [ 19]: Content-Length: 262 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 14 [ 0]: [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 0 [ 3]: v=0 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 1 [ 49]: o=root 1411874980 1411874980 IN IP4 192.168.10.70 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 5 [ 29]: m=audio 19452 RTP/AVP 8 0 101 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 11 [ 10]: a=sendrecv [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #383 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 16:37:03] DEBUG[2187] features.c: bridge answer set, chan answer set [Oct 22 16:37:03] DEBUG[2187] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 22 16:37:03] DEBUG[2187] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 22 16:37:03] VERBOSE[2187] rtp_engine.c: -- Remotely bridging SIP/phone1-00000016 and SIP/phone2-00000017 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Deferring reinvite on SIP '3c26c3d24efb-50v3t7e4o9ys' - It's audio will be redirected to IP 192.168.10.204:52688 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Sending reinvite on SIP '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' - It's audio soon redirected to IP 192.168.10.203:56056 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Strict routing enforced for session 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:03] VERBOSE[2187] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 22 16:37:03] DEBUG[2187] netsock2.c: Splitting '192.168.10.204:2050' gives... [Oct 22 16:37:03] DEBUG[2187] netsock2.c: ...host '192.168.10.204' and port '2050'. [Oct 22 16:37:03] VERBOSE[2187] chan_sip.c: set_destination: set destination to 192.168.10.204:2050 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 16:37:03] VERBOSE[2187] chan_sip.c: Audio is at 5060 [Oct 22 16:37:03] VERBOSE[2187] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Oct 22 16:37:03] VERBOSE[2187] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Oct 22 16:37:03] VERBOSE[2187] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Initializing already initialized SIP dialog 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 (presumably reinvite) [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.204:2050;line=xguiqus6 SIP/2.0 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK5bf933b4 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 4 [ 65]: To: ;tag=ryuq3pvs4i [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 5 [ 37]: Contact: [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 6 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Oct 22 16:37:03] VERBOSE[2187] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.204:2050: INVITE sip:phone2@192.168.10.204:2050;line=xguiqus6 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK5bf933b4 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as3c058282 To: ;tag=ryuq3pvs4i Contact: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Erika Musterfrau" Content-Type: application/sdp Content-Length: 264 v=0 o=root 1862472553 1862472554 IN IP4 192.168.10.203 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.203 t=0 0 m=audio 56056 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.204:2050;line=xguiqus6 SIP/2.0 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK5bf933b4 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 4 [ 65]: To: ;tag=ryuq3pvs4i [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 5 [ 37]: Contact: [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 6 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 14 [ 19]: Content-Length: 264 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Header 15 [ 0]: [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 0 [ 3]: v=0 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 1 [ 50]: o=root 1862472553 1862472554 IN IP4 192.168.10.203 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.203 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 5 [ 29]: m=audio 56056 RTP/AVP 8 0 101 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Body 11 [ 10]: a=sendrecv [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #384 [Oct 22 16:37:03] DEBUG[2187] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 16:37:03] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 16:37:03] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 16:37:03] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 16:37:03] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 16:37:03] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 16:37:03] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 16:37:03] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 16:37:03] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 16:37:03] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-00000017 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 150 CallerIDName: Uniqueid: 1287758221.25 [Oct 22 16:37:03] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone1-00000016 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1287758221.24 [Oct 22 16:37:03] DEBUG[1854] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/phone2-00000017 Uniqueid: 1287758221.25 AccountCode: OldAccountCode: [Oct 22 16:37:03] DEBUG[1854] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/phone1-00000016 Channel2: SIP/phone2-00000017 Uniqueid1: 1287758221.24 Uniqueid2: 1287758221.25 CallerID1: 100 CallerID2: 150 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Method: INVITE [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c26c3d24efb-50v3t7e4o9ys' Method: INVITE [Oct 22 16:37:03] DEBUG[1846] app_queue.c: Extension '150@Standard' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 22 16:37:03] DEBUG[1872] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 22 16:37:03] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 1 [Oct 22 16:37:03] DEBUG[1872] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 22 16:37:03] DEBUG[1872] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.203:2051 ---> ACK sip:150@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.203:2051;branch=z9hG4bK-s2zzh9upm18c;rport From: "Erika" ;tag=3vqvxksyq8 To: ;tag=as388f7611 Call-ID: 3c26c3d24efb-50v3t7e4o9ys CSeq: 1 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 0 [ 38]: ACK sip:150@192.168.10.70:5060 SIP/2.0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.10.203:2051;branch=z9hG4bK-s2zzh9upm18c;rport [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 2 [ 55]: From: "Erika" ;tag=3vqvxksyq8 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 3 [ 53]: To: ;tag=as388f7611 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 4 [ 34]: Call-ID: 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 7 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: --- (9 headers 0 lines) --- [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: = Looking for Call ID: 3c26c3d24efb-50v3t7e4o9ys (Checking From) --From tag 3vqvxksyq8 --To-tag as388f7611 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #383 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c26c3d24efb-50v3t7e4o9ys' of Response 1: Match Found [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Sending pending reinvite on '3c26c3d24efb-50v3t7e4o9ys' [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Strict routing enforced for session 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 22 16:37:03] DEBUG[1852] netsock2.c: Splitting '192.168.10.203:2051' gives... [Oct 22 16:37:03] DEBUG[1852] netsock2.c: ...host '192.168.10.203' and port '2051'. [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: set_destination: set destination to 192.168.10.203:2051 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Audio is at 5060 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Initializing already initialized SIP dialog 3c26c3d24efb-50v3t7e4o9ys (presumably reinvite) [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.203:2051;line=4db7r649 SIP/2.0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK20c5c5f2;rport [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 3 [ 55]: From: ;tag=as388f7611 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 4 [ 53]: To: "Erika" ;tag=3vqvxksyq8 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 5 [ 37]: Contact: [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 6 [ 34]: Call-ID: 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 9 [ 14]: Require: timer [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 12 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 14 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 15 [ 58]: P-Asserted-Identity: "Hans Muster" [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 16 [ 29]: Content-Type: application/sdp [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.203:2051: INVITE sip:phone1@192.168.10.203:2051;line=4db7r649 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK20c5c5f2;rport Max-Forwards: 70 From: ;tag=as388f7611 To: "Erika" ;tag=3vqvxksyq8 Contact: Call-ID: 3c26c3d24efb-50v3t7e4o9ys CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.0-1 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Hans Muster" Content-Type: application/sdp Content-Length: 264 v=0 o=root 1411874980 1411874981 IN IP4 192.168.10.204 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.204 t=0 0 m=audio 52688 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.203:2051;line=4db7r649 SIP/2.0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK20c5c5f2;rport [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 3 [ 55]: From: ;tag=as388f7611 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 4 [ 53]: To: "Erika" ;tag=3vqvxksyq8 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 5 [ 37]: Contact: [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 6 [ 34]: Call-ID: 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 9 [ 14]: Require: timer [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 12 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 14 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 15 [ 58]: P-Asserted-Identity: "Hans Muster" [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 16 [ 29]: Content-Type: application/sdp [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 17 [ 19]: Content-Length: 264 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 18 [ 0]: [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 0 [ 3]: v=0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 1 [ 50]: o=root 1411874980 1411874981 IN IP4 192.168.10.204 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.204 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 5 [ 29]: m=audio 52688 RTP/AVP 8 0 101 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 11 [ 10]: a=sendrecv [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #385 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Method: INVITE [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c26c3d24efb-50v3t7e4o9ys' Method: ACK [Oct 22 16:37:03] DEBUG[2187] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Oct 22 16:37:03] DEBUG[1854] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.203:56057 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 16777216 FractionLost: 0 PacketsLost: 1 HighestSequence: 0 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.204:2050 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK5bf933b4 From: "Erika Musterfrau" ;tag=as3c058282 To: ;tag=ryuq3pvs4i Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 103 INVITE Contact: ;reg-id=1 User-Agent: snom320/7.3.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 246 v=0 o=root 514665578 514665580 IN IP4 192.168.10.204 s=call c=IN IP4 192.168.10.204 t=0 0 m=audio 52688 RTP/AVP 8 0 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK5bf933b4 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 3 [ 65]: To: ;tag=ryuq3pvs4i [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 4 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 7 [ 26]: User-Agent: snom320/7.3.30 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 8 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 9 [ 42]: Allow-Events: talk, hold, refer, call-info [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 10 [ 47]: Supported: timer, 100rel, replaces, from-change [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 12 [ 19]: Content-Length: 246 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 13 [ 0]: [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 0 [ 3]: v=0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 1 [ 48]: o=root 514665578 514665580 IN IP4 192.168.10.204 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 2 [ 6]: s=call [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.204 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 5 [ 29]: m=audio 52688 RTP/AVP 8 0 101 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 pcma/8000 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 pcmu/8000 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 11 [ 10]: a=sendrecv [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: --- (13 headers 12 lines) --- [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: = Looking for Call ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 (Checking To) --From tag as3c058282 --To-tag ryuq3pvs4i [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Acked pending invite 103 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #384 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Stopping retransmission on '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' of Request 103: Match Found [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 514665578 514665580 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 16:37:03] DEBUG[1852] netsock2.c: Splitting '192.168.10.204' gives... [Oct 22 16:37:03] DEBUG[1852] netsock2.c: ...host '192.168.10.204' and port '(null)'. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found RTP audio format 8 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found RTP audio format 0 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found RTP audio format 101 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found audio description format pcma for ID 8 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found audio description format pcmu for ID 0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 22 16:37:03] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcc440d0' [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Peer audio RTP is at port 192.168.10.204:52688 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcc4427c [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcc4427c [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcc4427c [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 16:37:03] DEBUG[1852] netsock2.c: Splitting '192.168.10.204:2050' gives... [Oct 22 16:37:03] DEBUG[1852] netsock2.c: ...host '192.168.10.204' and port '2050'. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Strict routing enforced for session 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 22 16:37:03] DEBUG[1852] netsock2.c: Splitting '192.168.10.204:2050' gives... [Oct 22 16:37:03] DEBUG[1852] netsock2.c: ...host '192.168.10.204' and port '2050'. [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: set_destination: set destination to 192.168.10.204:2050 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Transmitting (no NAT) to 192.168.10.204:2050: ACK sip:phone2@192.168.10.204:2050;line=xguiqus6 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4c420814 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as3c058282 To: ;tag=ryuq3pvs4i Contact: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 0 [ 56]: ACK sip:phone2@192.168.10.204:2050;line=xguiqus6 SIP/2.0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4c420814 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 4 [ 65]: To: ;tag=ryuq3pvs4i [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 5 [ 37]: Contact: [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 6 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 10 [ 0]: [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Method: INVITE [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c26c3d24efb-50v3t7e4o9ys' Method: ACK [Oct 22 16:37:03] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 16:37:03] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 16:37:03] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 16:37:03] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 16:37:03] DEBUG[1872] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.203:2051 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK20c5c5f2;rport=5060 From: ;tag=as388f7611 To: "Erika" ;tag=3vqvxksyq8 Call-ID: 3c26c3d24efb-50v3t7e4o9ys CSeq: 102 INVITE Contact: ;reg-id=1 Require: timer Session-Expires: 1800;refresher=uas User-Agent: snom360/7.3.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 246 v=0 o=root 888790008 888790009 IN IP4 192.168.10.203 s=call c=IN IP4 192.168.10.203 t=0 0 m=audio 56056 RTP/AVP 8 0 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK20c5c5f2;rport=5060 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 2 [ 55]: From: ;tag=as388f7611 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 3 [ 53]: To: "Erika" ;tag=3vqvxksyq8 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 4 [ 34]: Call-ID: 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 7 [ 14]: Require: timer [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 8 [ 35]: Session-Expires: 1800;refresher=uas [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 9 [ 26]: User-Agent: snom360/7.3.30 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 10 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 14 [ 19]: Content-Length: 246 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 15 [ 0]: [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 0 [ 3]: v=0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 1 [ 48]: o=root 888790008 888790009 IN IP4 192.168.10.203 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 2 [ 6]: s=call [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.203 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 5 [ 29]: m=audio 56056 RTP/AVP 8 0 101 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 pcma/8000 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 pcmu/8000 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Body 11 [ 10]: a=sendrecv [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: --- (15 headers 12 lines) --- [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: = Looking for Call ID: 3c26c3d24efb-50v3t7e4o9ys (Checking To) --From tag as388f7611 --To-tag 3vqvxksyq8 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Acked pending invite 102 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #385 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c26c3d24efb-50v3t7e4o9ys' of Request 102: Match Found [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 888790008 888790009 IN IP4 192.168.10.203... UNSUPPORTED. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 16:37:03] DEBUG[1852] netsock2.c: Splitting '192.168.10.203' gives... [Oct 22 16:37:03] DEBUG[1852] netsock2.c: ...host '192.168.10.203' and port '(null)'. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found RTP audio format 8 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found RTP audio format 0 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found RTP audio format 101 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found audio description format pcma for ID 8 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found audio description format pcmu for ID 0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 22 16:37:03] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd270a0' [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Peer audio RTP is at port 192.168.10.203:56056 [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd2724c [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd2724c [Oct 22 16:37:03] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd2724c [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Updating call counter for incoming call [Oct 22 16:37:03] DEBUG[1852] netsock2.c: Splitting '192.168.10.203:2051' gives... [Oct 22 16:37:03] DEBUG[1852] netsock2.c: ...host '192.168.10.203' and port '2051'. [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Strict routing enforced for session 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 22 16:37:03] DEBUG[1852] netsock2.c: Splitting '192.168.10.203:2051' gives... [Oct 22 16:37:03] DEBUG[1852] netsock2.c: ...host '192.168.10.203' and port '2051'. [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: set_destination: set destination to 192.168.10.203:2051 [Oct 22 16:37:03] VERBOSE[1852] chan_sip.c: Transmitting (no NAT) to 192.168.10.203:2051: ACK sip:phone1@192.168.10.203:2051;line=4db7r649 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK391b8d88;rport Max-Forwards: 70 From: ;tag=as388f7611 To: "Erika" ;tag=3vqvxksyq8 Contact: Call-ID: 3c26c3d24efb-50v3t7e4o9ys CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 0 [ 56]: ACK sip:phone1@192.168.10.203:2051;line=4db7r649 SIP/2.0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK391b8d88;rport [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 3 [ 55]: From: ;tag=as388f7611 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 4 [ 53]: To: "Erika" ;tag=3vqvxksyq8 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 5 [ 37]: Contact: [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 6 [ 34]: Call-ID: 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Header 10 [ 0]: [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Method: INVITE [Oct 22 16:37:03] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c26c3d24efb-50v3t7e4o9ys' Method: ACK [Oct 22 16:37:03] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 16:37:03] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 16:37:03] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 16:37:03] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 16:37:03] DEBUG[1872] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 22 16:37:04] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Method: INVITE [Oct 22 16:37:04] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c26c3d24efb-50v3t7e4o9ys' Method: ACK [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.204:2050 ---> INVITE sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-369wprwwuorh;rport From: ;tag=ryuq3pvs4i To: "Erika Musterfrau" ;tag=as3c058282 Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;reg-id=1 P-Key-Flags: keys="3" User-Agent: snom320/7.3.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 246 v=0 o=root 514665578 514665581 IN IP4 192.168.10.204 s=call c=IN IP4 192.168.10.204 t=0 0 m=audio 52688 RTP/AVP 8 0 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendonly <-------------> [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 0 [ 41]: INVITE sip:100@192.168.10.70:5060 SIP/2.0 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-369wprwwuorh;rport [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 2 [ 67]: From: ;tag=ryuq3pvs4i [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 4 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 7 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 8 [ 21]: P-Key-Flags: keys="3" [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 9 [ 26]: User-Agent: snom320/7.3.30 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 11 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 12 [ 42]: Allow-Events: talk, hold, refer, call-info [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 13 [ 47]: Supported: timer, 100rel, replaces, from-change [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 14 [ 35]: Session-Expires: 3600;refresher=uas [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 15 [ 10]: Min-SE: 90 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 16 [ 29]: Content-Type: application/sdp [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 17 [ 19]: Content-Length: 246 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 18 [ 0]: [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 0 [ 3]: v=0 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 1 [ 48]: o=root 514665578 514665581 IN IP4 192.168.10.204 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 2 [ 6]: s=call [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.204 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 5 [ 29]: m=audio 52688 RTP/AVP 8 0 101 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 pcma/8000 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 pcmu/8000 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 11 [ 10]: a=sendonly [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: --- (18 headers 12 lines) --- [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: = Looking for Call ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 (Checking From) --From tag ryuq3pvs4i --To-tag as3c058282 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Oct 22 16:37:05] DEBUG[1852] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change" [Oct 22 16:37:05] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -timer- [Oct 22 16:37:05] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: timer [Oct 22 16:37:05] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -100rel- [Oct 22 16:37:05] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: 100rel [Oct 22 16:37:05] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -replaces- [Oct 22 16:37:05] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: replaces [Oct 22 16:37:05] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -from-change- [Oct 22 16:37:05] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: from-change [Oct 22 16:37:05] DEBUG[1852] netsock2.c: Splitting '192.168.10.204:2050' gives... [Oct 22 16:37:05] DEBUG[1852] netsock2.c: ...host '192.168.10.204' and port '2050'. [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Sending to 192.168.10.204:2050 (no NAT) [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Initializing initreq for method INVITE - callid 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 514665578 514665581 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 16:37:05] DEBUG[1852] netsock2.c: Splitting '192.168.10.204' gives... [Oct 22 16:37:05] DEBUG[1852] netsock2.c: ...host '192.168.10.204' and port '(null)'. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Found RTP audio format 8 [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33a9f88 [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Found RTP audio format 0 [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33a9f88 [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Found RTP audio format 101 [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33a9f88 [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Found audio description format pcma for ID 8 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Found audio description format pcmu for ID 0 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33a9f88 [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33a9f88 [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33a9f88 [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 22 16:37:05] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcc440d0' [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Peer audio RTP is at port 192.168.10.204:52688 [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33a9f88 to 0xcc4427c [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33a9f88 to 0xcc4427c [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33a9f88 to 0xcc4427c [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 16:37:05] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcc440d0' [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Got a SIP re-invite for call 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Session-Expires: 3600 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Refresher: UAS [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Received Min-SE: 90 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: SIP/phone2-00000017: This call is UP.... [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.204:2050 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-369wprwwuorh;received=192.168.10.204;rport=2050 From: ;tag=ryuq3pvs4i To: "Erika Musterfrau" ;tag=as3c058282 Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 1 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-369wprwwuorh;received=192.168.10.204;rport=2050 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 2 [ 67]: From: ;tag=ryuq3pvs4i [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 4 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 9 [ 37]: Contact: [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 11 [ 0]: [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Setting framing from config on incoming call [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Audio is at 5060 [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.204:2050 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-369wprwwuorh;received=192.168.10.204;rport=2050 From: ;tag=ryuq3pvs4i To: "Erika Musterfrau" ;tag=as3c058282 Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 1 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 1862472553 1862472555 IN IP4 192.168.10.203 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.203 t=0 0 m=audio 56056 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly <------------> [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-369wprwwuorh;received=192.168.10.204;rport=2050 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 2 [ 67]: From: ;tag=ryuq3pvs4i [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 4 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 9 [ 37]: Contact: [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 11 [ 19]: Content-Length: 264 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 12 [ 0]: [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 0 [ 3]: v=0 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 1 [ 50]: o=root 1862472553 1862472555 IN IP4 192.168.10.203 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.203 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 5 [ 29]: m=audio 56056 RTP/AVP 8 0 101 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 11 [ 10]: a=recvonly [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #386 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Method: INVITE [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c26c3d24efb-50v3t7e4o9ys' Method: ACK [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Sending reinvite on SIP '3c26c3d24efb-50v3t7e4o9ys' - It's audio soon redirected to IP 192.168.10.70:5060 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Strict routing enforced for session 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:05] VERBOSE[2187] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 22 16:37:05] DEBUG[2187] netsock2.c: Splitting '192.168.10.203:2051' gives... [Oct 22 16:37:05] DEBUG[2187] netsock2.c: ...host '192.168.10.203' and port '2051'. [Oct 22 16:37:05] VERBOSE[2187] chan_sip.c: set_destination: set destination to 192.168.10.203:2051 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Oct 22 16:37:05] VERBOSE[2187] chan_sip.c: Audio is at 5060 [Oct 22 16:37:05] VERBOSE[2187] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Oct 22 16:37:05] VERBOSE[2187] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Oct 22 16:37:05] VERBOSE[2187] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Initializing already initialized SIP dialog 3c26c3d24efb-50v3t7e4o9ys (presumably reinvite) [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.203:2051;line=4db7r649 SIP/2.0 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK03075c5f;rport [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 3 [ 55]: From: ;tag=as388f7611 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 4 [ 53]: To: "Erika" ;tag=3vqvxksyq8 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 5 [ 37]: Contact: [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 6 [ 34]: Call-ID: 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 9 [ 14]: Require: timer [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 12 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 14 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 15 [ 58]: P-Asserted-Identity: "Hans Muster" [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 16 [ 29]: Content-Type: application/sdp [Oct 22 16:37:05] VERBOSE[2187] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.203:2051: INVITE sip:phone1@192.168.10.203:2051;line=4db7r649 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK03075c5f;rport Max-Forwards: 70 From: ;tag=as388f7611 To: "Erika" ;tag=3vqvxksyq8 Contact: Call-ID: 3c26c3d24efb-50v3t7e4o9ys CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.0-1 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Hans Muster" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1411874980 1411874982 IN IP4 192.168.10.70 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.70 t=0 0 m=audio 19452 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.203:2051;line=4db7r649 SIP/2.0 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK03075c5f;rport [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 3 [ 55]: From: ;tag=as388f7611 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 4 [ 53]: To: "Erika" ;tag=3vqvxksyq8 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 5 [ 37]: Contact: [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 6 [ 34]: Call-ID: 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 9 [ 14]: Require: timer [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 12 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 14 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 15 [ 58]: P-Asserted-Identity: "Hans Muster" [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 16 [ 29]: Content-Type: application/sdp [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 17 [ 19]: Content-Length: 262 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Header 18 [ 0]: [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Body 0 [ 3]: v=0 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Body 1 [ 49]: o=root 1411874980 1411874982 IN IP4 192.168.10.70 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Body 5 [ 29]: m=audio 19452 RTP/AVP 8 0 101 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Body 11 [ 10]: a=sendrecv [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #387 [Oct 22 16:37:05] DEBUG[2187] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 16:37:05] DEBUG[2187] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 22 16:37:05] VERBOSE[2187] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/phone1-00000016 [Oct 22 16:37:05] DEBUG[2187] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 22 16:37:05] DEBUG[2187] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Method: INVITE [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c26c3d24efb-50v3t7e4o9ys' Method: ACK [Oct 22 16:37:05] DEBUG[1854] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/phone1-00000016 UniqueID: 1287758221.24 Class: default [Oct 22 16:37:05] DEBUG[2187] channel.c: Set channel SIP/phone1-00000016 to write format slin [Oct 22 16:37:05] DEBUG[2187] res_musiconhold.c: SIP/phone1-00000016 Opened file 0 '/var/lib/asterisk/moh/iptam/iptam_moh' [Oct 22 16:37:05] DEBUG[2187] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Oct 22 16:37:05] DEBUG[2187] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Oct 22 16:37:05] DEBUG[2187] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xcd270a0' [Oct 22 16:37:05] DEBUG[2187] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Oct 22 16:37:05] DEBUG[1854] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.203:56057 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 1663566077 FractionLost: 253 PacketsLost: 10339 HighestSequence: 58302 SequenceNumberCycles: 0 IAJitter: 10055529 LastSR: 8207.2147483648 DLSR: 1.9360(sec) RTT: 149(sec) [Oct 22 16:37:05] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.203:2051 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK03075c5f;rport=5060 From: ;tag=as388f7611 To: "Erika" ;tag=3vqvxksyq8 Call-ID: 3c26c3d24efb-50v3t7e4o9ys CSeq: 103 INVITE Contact: ;reg-id=1 Require: timer Session-Expires: 1800;refresher=uas User-Agent: snom360/7.3.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 246 v=0 o=root 888790008 888790010 IN IP4 192.168.10.203 s=call c=IN IP4 192.168.10.203 t=0 0 m=audio 56056 RTP/AVP 8 0 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK03075c5f;rport=5060 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 2 [ 55]: From: ;tag=as388f7611 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 3 [ 53]: To: "Erika" ;tag=3vqvxksyq8 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 4 [ 34]: Call-ID: 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 7 [ 14]: Require: timer [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 8 [ 35]: Session-Expires: 1800;refresher=uas [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 9 [ 26]: User-Agent: snom360/7.3.30 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 10 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 14 [ 19]: Content-Length: 246 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 15 [ 0]: [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 0 [ 3]: v=0 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 1 [ 48]: o=root 888790008 888790010 IN IP4 192.168.10.203 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 2 [ 6]: s=call [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.203 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 5 [ 29]: m=audio 56056 RTP/AVP 8 0 101 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 pcma/8000 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 pcmu/8000 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Body 11 [ 10]: a=sendrecv [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: --- (15 headers 12 lines) --- [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: = Looking for Call ID: 3c26c3d24efb-50v3t7e4o9ys (Checking To) --From tag as388f7611 --To-tag 3vqvxksyq8 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Acked pending invite 103 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #387 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c26c3d24efb-50v3t7e4o9ys' of Request 103: Match Found [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 888790008 888790010 IN IP4 192.168.10.203... UNSUPPORTED. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 16:37:05] DEBUG[1852] netsock2.c: Splitting '192.168.10.203' gives... [Oct 22 16:37:05] DEBUG[1852] netsock2.c: ...host '192.168.10.203' and port '(null)'. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Found RTP audio format 8 [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Found RTP audio format 0 [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Found RTP audio format 101 [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Found audio description format pcma for ID 8 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Found audio description format pcmu for ID 0 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 22 16:37:05] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd270a0' [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Peer audio RTP is at port 192.168.10.203:56056 [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd2724c [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd2724c [Oct 22 16:37:05] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd2724c [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Updating call counter for incoming call [Oct 22 16:37:05] DEBUG[1852] netsock2.c: Splitting '192.168.10.203:2051' gives... [Oct 22 16:37:05] DEBUG[1852] netsock2.c: ...host '192.168.10.203' and port '2051'. [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Strict routing enforced for session 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 22 16:37:05] DEBUG[1852] netsock2.c: Splitting '192.168.10.203:2051' gives... [Oct 22 16:37:05] DEBUG[1852] netsock2.c: ...host '192.168.10.203' and port '2051'. [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: set_destination: set destination to 192.168.10.203:2051 [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: Transmitting (no NAT) to 192.168.10.203:2051: ACK sip:phone1@192.168.10.203:2051;line=4db7r649 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK35925297;rport Max-Forwards: 70 From: ;tag=as388f7611 To: "Erika" ;tag=3vqvxksyq8 Contact: Call-ID: 3c26c3d24efb-50v3t7e4o9ys CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 0 [ 56]: ACK sip:phone1@192.168.10.203:2051;line=4db7r649 SIP/2.0 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK35925297;rport [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 3 [ 55]: From: ;tag=as388f7611 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 4 [ 53]: To: "Erika" ;tag=3vqvxksyq8 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 5 [ 37]: Contact: [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 6 [ 34]: Call-ID: 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 10 [ 0]: [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Method: INVITE [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c26c3d24efb-50v3t7e4o9ys' Method: ACK [Oct 22 16:37:05] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:05] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 16:37:05] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 16:37:05] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 16:37:05] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.204:2050 ---> ACK sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-brkcn6mgkemr;rport From: ;tag=ryuq3pvs4i To: "Erika Musterfrau" ;tag=as3c058282 Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 1 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 0 [ 38]: ACK sip:100@192.168.10.70:5060 SIP/2.0 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-brkcn6mgkemr;rport [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 2 [ 67]: From: ;tag=ryuq3pvs4i [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 4 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 7 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Oct 22 16:37:05] VERBOSE[1852] chan_sip.c: --- (9 headers 0 lines) --- [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: = Looking for Call ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 (Checking From) --From tag ryuq3pvs4i --To-tag as3c058282 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #386 [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Stopping retransmission on '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' of Response 1: Match Found [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Method: ACK [Oct 22 16:37:05] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c26c3d24efb-50v3t7e4o9ys' Method: ACK [Oct 22 16:37:05] DEBUG[1872] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 22 16:37:05] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:05] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:05] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:05] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:05] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:05] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:05] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Method: ACK [Oct 22 16:37:06] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c26c3d24efb-50v3t7e4o9ys' Method: ACK [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:06] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Method: ACK [Oct 22 16:37:07] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c26c3d24efb-50v3t7e4o9ys' Method: ACK [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:07] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.204:2050 ---> REFER sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-g548twiwmscb;rport From: ;tag=ryuq3pvs4i To: "Erika Musterfrau" ;tag=as3c058282 Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 2 REFER Max-Forwards: 70 Contact: ;reg-id=1 Refer-To: sip:180@192.168.10.70;user=phone Referred-By: sip:phone2@192.168.10.70 User-Agent: snom320/7.3.30 Content-Length: 0 <-------------> [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 0 [ 40]: REFER sip:100@192.168.10.70:5060 SIP/2.0 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-g548twiwmscb;rport [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 2 [ 67]: From: ;tag=ryuq3pvs4i [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 4 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 5 [ 13]: CSeq: 2 REFER [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 7 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 8 [ 42]: Refer-To: sip:180@192.168.10.70;user=phone [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 9 [ 37]: Referred-By: sip:phone2@192.168.10.70 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 10 [ 26]: User-Agent: snom320/7.3.30 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: --- (12 headers 0 lines) --- [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: = Looking for Call ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 (Checking From) --From tag ryuq3pvs4i --To-tag as3c058282 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: Call 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 got a SIP call transfer from caller: (REFER)! [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: SIP transfer to extension 180@Standard by phone2@192.168.10.70 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: SIP blind transfer: Transferer channel SIP/phone2-00000017, transferee channel SIP/phone1-00000016 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Got SIP transfer, applying to bridged peer 'SIP/phone1-00000016' [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.204:2050 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-g548twiwmscb;received=192.168.10.204;rport=2050 From: ;tag=ryuq3pvs4i To: "Erika Musterfrau" ;tag=as3c058282 Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 2 REFER Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 0 [ 20]: SIP/2.0 202 Accepted [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-g548twiwmscb;received=192.168.10.204;rport=2050 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 2 [ 67]: From: ;tag=ryuq3pvs4i [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 4 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 5 [ 13]: CSeq: 2 REFER [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 9 [ 37]: Contact: [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 11 [ 0]: [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: chan1->name: SIP/phone2-00000017 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Strict routing enforced for session 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 22 16:37:08] DEBUG[1852] netsock2.c: Splitting '192.168.10.204:2050' gives... [Oct 22 16:37:08] DEBUG[1852] netsock2.c: ...host '192.168.10.204' and port '2050'. [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: set_destination: set destination to 192.168.10.204:2050 [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.204:2050: NOTIFY sip:phone2@192.168.10.204:2050;line=xguiqus6 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4ea18348;rport Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as3c058282 To: ;tag=ryuq3pvs4i Contact: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 104 NOTIFY User-Agent: Asterisk PBX 1.8.0-1 Event: refer;id=2 Subscription-state: active Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 21 SIP/2.0 183 Ringing --- [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 0 [ 59]: NOTIFY sip:phone2@192.168.10.204:2050;line=xguiqus6 SIP/2.0 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4ea18348;rport [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 4 [ 65]: To: ;tag=ryuq3pvs4i [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 5 [ 37]: Contact: [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 6 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 7 [ 16]: CSeq: 104 NOTIFY [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 9 [ 17]: Event: refer;id=2 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 10 [ 26]: Subscription-state: active [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 11 [ 41]: Content-Type: message/sipfrag;version=2.0 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 12 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 14 [ 18]: Content-Length: 21 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 15 [ 0]: [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Body 0 [ 19]: SIP/2.0 183 Ringing [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #389 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 16:37:08] DEBUG[1852] channel.c: Soft-Hanging up channel 'SIP/phone1-00000016' [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Blind transfer succeeded. Telling transferer. [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Strict routing enforced for session 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 22 16:37:08] DEBUG[1852] netsock2.c: Splitting '192.168.10.204:2050' gives... [Oct 22 16:37:08] DEBUG[1852] netsock2.c: ...host '192.168.10.204' and port '2050'. [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: set_destination: set destination to 192.168.10.204:2050 [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.204:2050: NOTIFY sip:phone2@192.168.10.204:2050;line=xguiqus6 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1997c0d8;rport Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as3c058282 To: ;tag=ryuq3pvs4i Contact: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 105 NOTIFY User-Agent: Asterisk PBX 1.8.0-1 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, INFO, PUBLISH Supported: replaces, timer Content-Length: 16 SIP/2.0 200 Ok --- [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 0 [ 59]: NOTIFY sip:phone2@192.168.10.204:2050;line=xguiqus6 SIP/2.0 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1997c0d8;rport [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 4 [ 65]: To: ;tag=ryuq3pvs4i [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 5 [ 37]: Contact: [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 6 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 7 [ 16]: CSeq: 105 NOTIFY [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 9 [ 17]: Event: refer;id=2 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 10 [ 48]: Subscription-state: terminated;reason=noresource [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 11 [ 41]: Content-Type: message/sipfrag;version=2.0 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 12 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 14 [ 18]: Content-Length: 16 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 15 [ 0]: [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Body 0 [ 14]: SIP/2.0 200 Ok [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #390 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Method: REFER [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c26c3d24efb-50v3t7e4o9ys' Method: ACK [Oct 22 16:37:08] DEBUG[1854] manager.c: Examining event: Event: Transfer Privilege: call,all TransferMethod: SIP TransferType: Blind Channel: SIP/phone2-00000017 Uniqueid: 1287758221.25 SIP-Callid: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 TargetChannel: SIP/phone1-00000016 TargetUniqueid: 1287758221.24 TransferExten: 180 TransferContext: Standard [Oct 22 16:37:08] VERBOSE[2187] res_musiconhold.c: -- Stopped music on hold on SIP/phone1-00000016 [Oct 22 16:37:08] DEBUG[2187] channel.c: Set channel SIP/phone1-00000016 to write format alaw [Oct 22 16:37:08] DEBUG[2187] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 22 16:37:08] DEBUG[2187] rtp_engine.c: Oooh, got a hangup [Oct 22 16:37:08] DEBUG[2187] channel.c: Returning from native bridge, channels: SIP/phone1-00000016, SIP/phone2-00000017 [Oct 22 16:37:08] DEBUG[2187] channel.c: Hanging up channel 'SIP/phone2-00000017' [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: update_call_counter(phone2) - decrement call limit counter on hangup [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Updating call counter for outgoing call [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Call to peer 'phone2' removed from call limit 2147483647 [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70. [Oct 22 16:37:08] VERBOSE[2187] chan_sip.c: Scheduling destruction of SIP dialog '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' in 32000 ms (Method: REFER) [Oct 22 16:37:08] DEBUG[2187] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Oct 22 16:37:08] DEBUG[2187] pbx.c: Spawn extension (Standard,180,1) exited non-zero on 'SIP/phone1-00000016' [Oct 22 16:37:08] VERBOSE[2187] pbx.c: == Spawn extension (Standard, 180, 1) exited non-zero on 'SIP/phone1-00000016' [Oct 22 16:37:08] DEBUG[2187] channel.c: Soft-Hanging up channel 'SIP/phone1-00000016' [Oct 22 16:37:08] DEBUG[2187] channel.c: Hanging up channel 'SIP/phone1-00000016' [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Hangup call SIP/phone1-00000016, SIP callid 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: update_call_counter(phone1) - decrement call limit counter on hangup [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Updating call counter for incoming call [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Call from peer 'phone1' removed from call limit 2147483647 [Oct 22 16:37:08] DEBUG[2187] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd270a0' [Oct 22 16:37:08] VERBOSE[2187] chan_sip.c: Scheduling destruction of SIP dialog '3c26c3d24efb-50v3t7e4o9ys' in 32000 ms (Method: ACK) [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Session timer stopped: -1 - 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Strict routing enforced for session 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:08] VERBOSE[2187] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 22 16:37:08] DEBUG[2187] netsock2.c: Splitting '192.168.10.203:2051' gives... [Oct 22 16:37:08] DEBUG[2187] netsock2.c: ...host '192.168.10.203' and port '2051'. [Oct 22 16:37:08] VERBOSE[2187] chan_sip.c: set_destination: set destination to 192.168.10.203:2051 [Oct 22 16:37:08] VERBOSE[2187] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.203:2051: BYE sip:phone1@192.168.10.203:2051;line=4db7r649 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK62e2833f;rport Max-Forwards: 70 From: ;tag=as388f7611 To: "Erika" ;tag=3vqvxksyq8 Call-ID: 3c26c3d24efb-50v3t7e4o9ys CSeq: 104 BYE User-Agent: Asterisk PBX 1.8.0-1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Header 0 [ 56]: BYE sip:phone1@192.168.10.203:2051;line=4db7r649 SIP/2.0 [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK62e2833f;rport [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Header 3 [ 55]: From: ;tag=as388f7611 [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Header 4 [ 53]: To: "Erika" ;tag=3vqvxksyq8 [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Header 5 [ 34]: Call-ID: 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Header 6 [ 13]: CSeq: 104 BYE [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Header 8 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Header 9 [ 30]: X-Asterisk-HangupCauseCode: 16 [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Header 11 [ 0]: [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #393 [Oct 22 16:37:08] DEBUG[2187] chan_sip.c: Trying to put 'BYE sip:pho' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 16:37:08] DEBUG[1854] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/phone1-00000016 Channel2: SIP/phone2-00000017 Uniqueid1: 1287758221.24 Uniqueid2: 1287758221.25 CallerID1: 100 CallerID2: 150 [Oct 22 16:37:08] DEBUG[1854] manager.c: Examining event: Event: Cdr Privilege: cdr,all AccountCode: Source: 100 Destination: 150 DestinationContext: Standard CallerID: "Erika Musterfrau" <100> Channel: SIP/phone1-00000016 DestinationChannel: SIP/phone2-00000017 LastApplication: Dial LastData: SIP/phone2 StartTime: 2010-10-22 16:37:01 AnswerTime: 2010-10-22 16:37:03 EndTime: 2010-10-22 16:37:08 Duration: 7 BillableSeconds: 5 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1287758221.24 UserField: [Oct 22 16:37:08] DEBUG[1854] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone2-00000017 Uniqueid: 1287758221.25 CallerIDNum: 150 CallerIDName: Cause: 16 Cause-txt: Normal Clearing [Oct 22 16:37:08] DEBUG[1854] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/phone1-00000016 UniqueID: 1287758221.24 DialStatus: ANSWER [Oct 22 16:37:08] DEBUG[1854] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone1-00000016 Uniqueid: 1287758221.24 CallerIDNum: 100 CallerIDName: Erika Musterfrau Cause: 16 Cause-txt: Normal Clearing [Oct 22 16:37:08] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 16:37:08] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 16:37:08] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Oct 22 16:37:08] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '1' [Oct 22 16:37:08] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 16:37:08] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 16:37:08] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Oct 22 16:37:08] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '1' [Oct 22 16:37:08] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 16:37:08] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 16:37:08] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Oct 22 16:37:08] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '1' [Oct 22 16:37:08] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 16:37:08] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 16:37:08] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Oct 22 16:37:08] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '1' [Oct 22 16:37:08] DEBUG[1846] app_queue.c: Extension '150@Standard' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 22 16:37:08] DEBUG[1846] app_queue.c: Extension '100@Standard' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 22 16:37:08] DEBUG[1872] app_queue.c: Device 'SIP/phone2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 22 16:37:08] DEBUG[1872] app_queue.c: Device 'SIP/phone2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 22 16:37:08] DEBUG[1872] app_queue.c: Device 'SIP/phone1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 22 16:37:08] DEBUG[1872] app_queue.c: Device 'SIP/phone1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 22 16:37:08] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 0 [Oct 22 16:37:08] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: Standard Hint: SIP/phone1 Status: 0 [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.203:2051 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK62e2833f;rport=5060 From: ;tag=as388f7611 To: "Erika" ;tag=3vqvxksyq8 Call-ID: 3c26c3d24efb-50v3t7e4o9ys CSeq: 104 BYE Contact: ;reg-id=1 User-Agent: snom360/7.3.30 RTP-RxStat: Total_Rx_Pkts=243,Rx_Pkts=242,Rx_Pkts_Lost=10339,Remote_Rx_Pkts_Lost=1 RTP-TxStat: Total_Tx_Pkts=237,Tx_Pkts=237,Remote_Tx_Pkts=7 Content-Length: 0 <-------------> [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK62e2833f;rport=5060 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 2 [ 55]: From: ;tag=as388f7611 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 3 [ 53]: To: "Erika" ;tag=3vqvxksyq8 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 4 [ 34]: Call-ID: 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 7 [ 26]: User-Agent: snom360/7.3.30 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 8 [ 82]: RTP-RxStat: Total_Rx_Pkts=243,Rx_Pkts=242,Rx_Pkts_Lost=10339,Remote_Rx_Pkts_Lost=1 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 9 [ 58]: RTP-TxStat: Total_Tx_Pkts=237,Tx_Pkts=237,Remote_Tx_Pkts=7 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: --- (11 headers 0 lines) --- [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: = Looking for Call ID: 3c26c3d24efb-50v3t7e4o9ys (Checking To) --From tag as388f7611 --To-tag 3vqvxksyq8 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #393 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c26c3d24efb-50v3t7e4o9ys' of Request 104: Match Found [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Destroying SIP dialog 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: Really destroying SIP dialog '3c26c3d24efb-50v3t7e4o9ys' Method: ACK [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: ---------- SIP HISTORY for '3c26c3d24efb-50v3t7e4o9ys' [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: * SIP Call [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 001. Rx INVITE / 1 INVITE / sip:150@192.168.10.70;user=phone [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 002. NewChan Channel SIP/phone1-00000016 - from 3c26c3d24efb-50v3t7e4o9ys [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 003. TxResp SIP/2.0 / 1 INVITE - 100 Trying [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 004. TxResp SIP/2.0 / 1 INVITE - 180 Ringing [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 005. ConnectedLine Called party is now Hans Muster <150> [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 006. TxRespRel SIP/2.0 / 1 INVITE - 200 OK [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 007. Rx ACK / 1 ACK / sip:150@192.168.10.70:5060 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 008. ReInv Re-invite sent [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 009. TxReqRel INVITE / 102 INVITE - INVITE [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 010. Rx SIP/2.0 / 102 INVITE / 200 Ok [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 011. TxReq ACK / 102 ACK - ACK [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 012. ReInv Re-invite sent [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 013. TxReqRel INVITE / 103 INVITE - INVITE [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 014. Rx SIP/2.0 / 103 INVITE / 200 Ok [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 015. TxReq ACK / 103 ACK - ACK [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 016. Hangup Cause Normal Clearing [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 017. SchedDestroy 32000 ms [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 018. RTCPaudio Quality:ssrc=46501312;themssrc=0;lp=0;rxjitter=0.000000;rxcount [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 019. TxReqRel BYE / 104 BYE - BYE [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 020. Rx SIP/2.0 / 104 BYE / 200 OK [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: 021. NeedDestroy Setting needdestroy because received 200 response [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: ---------- END SIP HISTORY for '3c26c3d24efb-50v3t7e4o9ys' [Oct 22 16:37:08] DEBUG[1852] rtp_engine.c: Destroyed RTP instance '0xcd270a0' [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.204:2050 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4ea18348;rport=5060 From: "Erika Musterfrau" ;tag=as3c058282 To: ;tag=ryuq3pvs4i Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 104 NOTIFY Contact: ;reg-id=1 Content-Length: 0 <-------------> [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4ea18348;rport=5060 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 3 [ 65]: To: ;tag=ryuq3pvs4i [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 4 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 5 [ 16]: CSeq: 104 NOTIFY [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: --- (8 headers 0 lines) --- [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: = Looking for Call ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 (Checking To) --From tag as3c058282 --To-tag ryuq3pvs4i [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #389 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Stopping retransmission on '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' of Request 104: Match Found [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: SIP Response message for INCOMING dialog NOTIFY arrived [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.204:2050 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1997c0d8;rport=5060 From: "Erika Musterfrau" ;tag=as3c058282 To: ;tag=ryuq3pvs4i Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 105 NOTIFY Contact: ;reg-id=1 Content-Length: 0 <-------------> [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1997c0d8;rport=5060 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 3 [ 65]: To: ;tag=ryuq3pvs4i [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 4 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 5 [ 16]: CSeq: 105 NOTIFY [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: --- (8 headers 0 lines) --- [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: = Looking for Call ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 (Checking To) --From tag as3c058282 --To-tag ryuq3pvs4i [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #390 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Stopping retransmission on '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' of Request 105: Match Found [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: SIP Response message for INCOMING dialog NOTIFY arrived [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: <--- SIP read from UDP:192.168.10.204:2050 ---> BYE sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-3s7xhyyc9i8x;rport From: ;tag=ryuq3pvs4i To: "Erika Musterfrau" ;tag=as3c058282 Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 3 BYE Max-Forwards: 70 Contact: ;reg-id=1 User-Agent: snom320/7.3.30 RTP-RxStat: Total_Rx_Pkts=92,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=104,Tx_Pkts=0,Remote_Tx_Pkts=0 Content-Length: 0 <-------------> [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 0 [ 38]: BYE sip:100@192.168.10.70:5060 SIP/2.0 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-3s7xhyyc9i8x;rport [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 2 [ 67]: From: ;tag=ryuq3pvs4i [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 4 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 5 [ 11]: CSeq: 3 BYE [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 7 [ 64]: Contact: ;reg-id=1 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 8 [ 26]: User-Agent: snom320/7.3.30 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 9 [ 75]: RTP-RxStat: Total_Rx_Pkts=92,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 10 [ 56]: RTP-TxStat: Total_Tx_Pkts=104,Tx_Pkts=0,Remote_Tx_Pkts=0 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: --- (12 headers 0 lines) --- [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: = Looking for Call ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 (Checking From) --From tag ryuq3pvs4i --To-tag as3c058282 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Initializing initreq for method BYE - callid 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:08] DEBUG[1852] netsock2.c: Splitting '192.168.10.204:2050' gives... [Oct 22 16:37:08] DEBUG[1852] netsock2.c: ...host '192.168.10.204' and port '2050'. [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: Sending to 192.168.10.204:2050 (no NAT) [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Setting SIP_ALREADYGONE on dialog 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:08] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcc440d0' [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: Scheduling destruction of SIP dialog '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' in 32000 ms (Method: BYE) [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Received bye, no owner, selfdestruct soon. [Oct 22 16:37:08] VERBOSE[1852] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.204:2050 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-3s7xhyyc9i8x;received=192.168.10.204;rport=2050 From: ;tag=ryuq3pvs4i To: "Erika Musterfrau" ;tag=as3c058282 Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 CSeq: 3 BYE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.204:2050;branch=z9hG4bK-3s7xhyyc9i8x;received=192.168.10.204;rport=2050 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 2 [ 67]: From: ;tag=ryuq3pvs4i [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as3c058282 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 4 [ 55]: Call-ID: 554fa2c24d7b76631f8a4bdc50956097@192.168.10.70 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 5 [ 11]: CSeq: 3 BYE [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Header 10 [ 0]: [Oct 22 16:37:08] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 16:37:40] VERBOSE[1852] chan_sip.c: Really destroying SIP dialog '554fa2c24d7b76631f8a4bdc50956097@192.168.10.70' Method: BYE