Перевод звонка - короткие гудки при поднятии трубки.
102, через 10 сек. если не ответили на номер 101. Так вот, с номера 102 вызов уходит на 101 и аппарат звонит, но при поднятии трубки сразу короткие гудки. Есть второй канал - USB модем с симкой. Входящие вызовы распределены так же, но при этом при поднятии трубки на 101 всё нормально. Проблема именно с аналоговой городской линией.
Это распределение входящих с городской линии:
| Код: |
| exten => s,1,Dial(SIP/102,10,t) exten => s,n,Dial(SIP/101,10,t) exten => s,n,Dial(SIP/103,10,t) exten => s,n,HangUp() |
Ни чего замысловатого вроде нет. На номере 101 аппарат Yealink SIP-T20, на номере 102 - Fanvil BW210. Где покопать ещё?
Предварительно подозрение на кодеки.
_________________
http://zemlyakovmp.ru - Мой блог о VoIP, Linux, Asterisk.
Вот что происходит. С городской линии приходит на 102, этот номер не ответил, переводится автоматом на 101. На 101 номере происходит всего один звонок и некоторое время тишина. У вызывающего в трубке гудки. Потом звонок переходит на номер 103. Если же поднять трубку после первого звонка на 101 номере - сразу занято.
Звонил с сотового.
Отладка:
| Код: |
| [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #36044 [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 192.168.3.100:5060 [2012-12-26 13:51:53] VERBOSE[9850] chan_sip.c: Scheduling destruction of SIP dialog '7f036b1617b5d0b37157f8bc7b09b83a@192.168.3.10:5060' in 6400 ms (Method: INVITE) [2012-12-26 13:51:53] DEBUG[9850] app_dial.c: Exiting with DIALSTATUS=NOANSWER. [2012-12-26 13:51:53] DEBUG[1842] devicestate.c: No provider found, checking channel drivers for SIP - 102 [2012-12-26 13:51:53] DEBUG[9850] pbx.c: Launching 'Dial' [2012-12-26 13:51:53] DEBUG[1842] chan_sip.c: Checking device state for peer 102 [2012-12-26 13:51:53] VERBOSE[9850] pbx.c: -- Executing [s@pstn:2] Dial("DAHDI/1-1", "SIP/101,10,t") in new stack [2012-12-26 13:51:53] DEBUG[1842] devicestate.c: Changing state for SIP/102 - state 1 (Not in use) [2012-12-26 13:51:53] DEBUG[1842] devicestate.c: device 'SIP/102' state '1' [2012-12-26 13:51:53] DEBUG[1918] app_queue.c: Device 'SIP/102' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Allocating new SIP dialog for 5258f3a038b35ffa45297f096f09c368@192.168.3.10:5060 - INVITE (No RTP) [2012-12-26 13:51:53] DEBUG[9850] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x223cce8' [2012-12-26 13:51:53] DEBUG[9850] res_rtp_asterisk.c: Allocated port 11414 for RTP instance '0x223cce8' [2012-12-26 13:51:53] DEBUG[9850] rtp_engine.c: RTP instance '0x223cce8' is setup and ready to go [2012-12-26 13:51:53] DEBUG[9850] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x223cce8' [2012-12-26 13:51:53] VERBOSE[9850] netsock2.c: == Using SIP RTP TOS bits 184 [2012-12-26 13:51:53] VERBOSE[9850] netsock2.c: == Using SIP RTP CoS mark 5 [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Setting NAT on RTP to On [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: OBPROXY: Not applying OBproxy to this call [2012-12-26 13:51:53] DEBUG[9850] acl.c: For destination '192.168.3.104', our source address is '192.168.3.10'. [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.3.10:5060 [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: *** Our native formats are 0x4 (ulaw) [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: This channel will not be able to handle video. [2012-12-26 13:51:53] DEBUG[9850] rtp_engine.c: Can't find native functions for channel 'DAHDI/1-1' [2012-12-26 13:51:53] DEBUG[9850] rtp_engine.c: Seeded SDP of 'SIP/101-00000069' with that of 'DAHDI/1-1' [2012-12-26 13:51:53] DEBUG[9850] channel.c: Not copying variable DIALEDTIME. [2012-12-26 13:51:53] DEBUG[9850] channel.c: Not copying variable ANSWEREDTIME. [2012-12-26 13:51:53] DEBUG[9850] channel.c: Not copying variable DIALEDPEERNAME. [2012-12-26 13:51:53] DEBUG[9850] channel.c: Not copying variable DIALEDPEERNUMBER. [2012-12-26 13:51:53] DEBUG[9850] channel.c: Not copying variable DIALSTATUS. [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Outgoing Call for 101 [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Updating call counter for outgoing call [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Call to peer '101' is 1 out of 1 [2012-12-26 13:51:53] DEBUG[1842] devicestate.c: No provider found, checking channel drivers for SIP - 101 [2012-12-26 13:51:53] DEBUG[1842] chan_sip.c: Checking device state for peer 101 [2012-12-26 13:51:53] DEBUG[1842] devicestate.c: Changing state for SIP/101 - state 6 (Ringing) [2012-12-26 13:51:53] DEBUG[1842] devicestate.c: device 'SIP/101' state '6' [2012-12-26 13:51:53] DEBUG[1918] app_queue.c: Device 'SIP/101' changed to state '6' (Ringing) [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: False Text flag: False [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [2012-12-26 13:51:53] VERBOSE[9850] chan_sip.c: Audio is at 11414 [2012-12-26 13:51:53] VERBOSE[9850] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2012-12-26 13:51:53] VERBOSE[9850] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: -- Done with adding codecs to SDP [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Initializing initreq for method INVITE - callid 3f89999821936b5271d44c61398d948d@192.168.3.10:5060 [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Header 0 [ 41]: INVITE sip:101@192.168.3.104:5062 SIP/2.0 [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK34b1e061;rport [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as3fd12559 [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Header 4 [ 32]: To: [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Header 5 [ 41]: Contact: [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Header 6 [ 59]: Call-ID: 3f89999821936b5271d44c61398d948d@192.168.3.10:5060 [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Header 8 [ 22]: User-Agent: 0041762415 [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Header 9 [ 35]: Date: Wed, 26 Dec 2012 09:51:53 GMT [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [2012-12-26 13:51:53] VERBOSE[9850] chan_sip.c: Reliably Transmitting (NAT) to 192.168.3.104:5062: INVITE sip:101@192.168.3.104:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK34b1e061;rport Max-Forwards: 70 From: "asterisk" ;tag=as3fd12559 To: Contact: Call-ID: 3f89999821936b5271d44c61398d948d@192.168.3.10:5060 CSeq: 102 INVITE User-Agent: 0041762415 Date: Wed, 26 Dec 2012 09:51:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 235 |
и далее
| Код: |
| v=0 o=root 254797653 254797653 IN IP4 192.168.3.10 s=Asterisk PBX 1.8.17.0 c=IN IP4 192.168.3.10 t=0 0 m=audio 11414 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #36046 [2012-12-26 13:51:53] DEBUG[9850] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.3.104:5062 [2012-12-26 13:51:53] VERBOSE[9850] app_dial.c: -- Called SIP/101 [2012-12-26 13:51:53] DEBUG[9850] chan_dahdi.c: Requested indication 22 on channel DAHDI/1-1 [2012-12-26 13:51:53] DEBUG[9850] chan_dahdi.c: Requested indication 22 on channel DAHDI/1-1 [2012-12-26 13:51:53] DEBUG[1916] at_read.c: [mts] receive 12 byte, used 12, free 2036, read 0, write 12 [2012-12-26 13:51:53] DEBUG[1916] at_read.c: [mts] [ ^RSSI:10 ] [2012-12-26 13:51:53] DEBUG[9850] dsp.c: tone 1100, Ew=1.18E+05, Et=6.78E+07, s/n= 0.00 [2012-12-26 13:51:53] DEBUG[9850] res_rtp_asterisk.c: No remote address on RTP instance '0x223cce8' so dropping frame [2012-12-26 13:51:53] VERBOSE[1871] chan_sip.c: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK34b1e061;rport From: "asterisk" ;tag=as3fd12559 To: Call-ID: 3f89999821936b5271d44c61398d948d@192.168.3.10:5060 CSeq: 102 INVITE User-Agent: Yealink SIP-T20P 9.60.14.20 Content-Length: 0 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK34b1e061;rport [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as3fd12559 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 3 [ 32]: To: [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 4 [ 59]: Call-ID: 3f89999821936b5271d44c61398d948d@192.168.3.10:5060 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 6 [ 39]: User-Agent: Yealink SIP-T20P 9.60.14.20 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [2012-12-26 13:51:53] VERBOSE[1871] chan_sip.c: --- (8 headers 0 lines) --- [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: = Looking for Call ID: 3f89999821936b5271d44c61398d948d@192.168.3.10:5060 (Checking To) --From tag as3fd12559 --To-tag [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: *** SIP TIMER: Cancelling retransmission #36046 - INVITE (got response) [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3f89999821936b5271d44c61398d948d@192.168.3.10:5060' Request 102: Found [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: SIP response 100 to standard invite [2012-12-26 13:51:53] VERBOSE[1871] chan_sip.c: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK66297e04 From: "asterisk" ;tag=as5b8988dc To: ;tag=307031427 Call-ID: 7f036b1617b5d0b37157f8bc7b09b83a@192.168.3.10:5060 CSeq: 102 CANCEL Server: Voip Phone 1.0 Content-Length: 0 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK66297e04 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as5b8988dc [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 3 [ 46]: To: ;tag=307031427 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 4 [ 59]: Call-ID: 7f036b1617b5d0b37157f8bc7b09b83a@192.168.3.10:5060 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 5 [ 16]: CSeq: 102 CANCEL [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 6 [ 22]: Server: Voip Phone 1.0 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [2012-12-26 13:51:53] VERBOSE[1871] chan_sip.c: --- (8 headers 0 lines) --- [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: = Looking for Call ID: 7f036b1617b5d0b37157f8bc7b09b83a@192.168.3.10:5060 (Checking To) --From tag as5b8988dc --To-tag 307031427 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Acked pending invite 102 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #36044 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Stopping retransmission on '7f036b1617b5d0b37157f8bc7b09b83a@192.168.3.10:5060' of Request 102: Match Found [2012-12-26 13:51:53] DEBUG[9850] dsp.c: tone 1100, Ew=1.39E+05, Et=6.83E+07, s/n= 0.00 [2012-12-26 13:51:53] DEBUG[9850] res_rtp_asterisk.c: No remote address on RTP instance '0x223cce8' so dropping frame [2012-12-26 13:51:53] DEBUG[9850] dsp.c: tone 1100, Ew=1.55E+05, Et=6.58E+07, s/n= 0.00 [2012-12-26 13:51:53] DEBUG[9850] res_rtp_asterisk.c: No remote address on RTP instance '0x223cce8' so dropping frame [2012-12-26 13:51:53] DEBUG[9850] dsp.c: tone 1100, Ew=9.52E+04, Et=6.42E+07, s/n= 0.00 [2012-12-26 13:51:53] DEBUG[9850] res_rtp_asterisk.c: No remote address on RTP instance '0x223cce8' so dropping frame [2012-12-26 13:51:53] DEBUG[9850] dsp.c: tone 1100, Ew=3.39E+05, Et=6.14E+07, s/n= 0.01 [2012-12-26 13:51:53] DEBUG[9850] res_rtp_asterisk.c: No remote address on RTP instance '0x223cce8' so dropping frame [2012-12-26 13:51:53] DEBUG[9850] dsp.c: tone 1100, Ew=1.19E+05, Et=6.50E+07, s/n= 0.00 [2012-12-26 13:51:53] DEBUG[9850] res_rtp_asterisk.c: No remote address on RTP instance '0x223cce8' so dropping frame [2012-12-26 13:51:53] VERBOSE[1871] chan_sip.c: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK66297e04 From: "asterisk" ;tag=as5b8988dc To: ;tag=307031427 Call-ID: 7f036b1617b5d0b37157f8bc7b09b83a@192.168.3.10:5060 CSeq: 102 INVITE Server: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 0 [ 30]: SIP/2.0 487 Request Terminated [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK66297e04 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as5b8988dc [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 3 [ 46]: To: ;tag=307031427 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 4 [ 59]: Call-ID: 7f036b1617b5d0b37157f8bc7b09b83a@192.168.3.10:5060 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 6 [ 22]: Server: Voip Phone 1.0 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 7 [ 85]: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [2012-12-26 13:51:53] VERBOSE[1871] chan_sip.c: --- (9 headers 0 lines) --- [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: = Looking for Call ID: 7f036b1617b5d0b37157f8bc7b09b83a@192.168.3.10:5060 (Checking To) --From tag as5b8988dc --To-tag 307031427 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Stopping retransmission on '7f036b1617b5d0b37157f8bc7b09b83a@192.168.3.10:5060' of Request 102: Match Found [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: SIP response 487 to standard invite [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Strict routing enforced for session 7f036b1617b5d0b37157f8bc7b09b83a@192.168.3.10:5060 [2012-12-26 13:51:53] VERBOSE[1871] chan_sip.c: Transmitting (no NAT) to 192.168.3.100:5060: ACK sip:102@192.168.3.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK66297e04 Max-Forwards: 70 From: "asterisk" ;tag=as5b8988dc To: ;tag=307031427 Contact: Call-ID: 7f036b1617b5d0b37157f8bc7b09b83a@192.168.3.10:5060 CSeq: 102 ACK User-Agent: 0041762415 Content-Length: 0 --- [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Trying to put 'ACK sip:102' onto UDP socket destined for 192.168.3.100:5060 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Updating call counter for outgoing call [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Call to peer '102' removed from call limit 1 [2012-12-26 13:51:53] VERBOSE[1871] chan_sip.c: Scheduling destruction of SIP dialog '7f036b1617b5d0b37157f8bc7b09b83a@192.168.3.10:5060' in 6400 ms (Method: INVITE) [2012-12-26 13:51:53] DEBUG[1842] devicestate.c: No provider found, checking channel drivers for SIP - 102 [2012-12-26 13:51:53] DEBUG[1842] chan_sip.c: Checking device state for peer 102 [2012-12-26 13:51:53] DEBUG[1842] devicestate.c: Changing state for SIP/102 - state 1 (Not in use) [2012-12-26 13:51:53] DEBUG[1842] devicestate.c: device 'SIP/102' state '1' [2012-12-26 13:51:53] DEBUG[1918] app_queue.c: Device 'SIP/102' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2012-12-26 13:51:53] DEBUG[9850] dsp.c: tone 1100, Ew=7.65E+03, Et=6.70E+07, s/n= 0.00 [2012-12-26 13:51:53] DEBUG[9850] res_rtp_asterisk.c: No remote address on RTP instance '0x223cce8' so dropping frame [2012-12-26 13:51:53] DEBUG[9850] dsp.c: tone 1100, Ew=6.27E+04, Et=6.41E+07, s/n= 0.00 [2012-12-26 13:51:53] DEBUG[9850] res_rtp_asterisk.c: No remote address on RTP instance '0x223cce8' so dropping frame [2012-12-26 13:51:53] DEBUG[9850] dsp.c: tone 1100, Ew=1.68E+05, Et=6.50E+07, s/n= 0.00 [2012-12-26 13:51:53] DEBUG[9850] res_rtp_asterisk.c: No remote address on RTP instance '0x223cce8' so dropping frame [2012-12-26 13:51:53] VERBOSE[1871] chan_sip.c: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK34b1e061;rport From: "asterisk" ;tag=as3fd12559 To: ;tag=819601190 Call-ID: 3f89999821936b5271d44c61398d948d@192.168.3.10:5060 CSeq: 102 INVITE Contact: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE User-Agent: Yealink SIP-T20P 9.60.14.20 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 0 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK34b1e061;rport [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as3fd12559 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 3 [ 46]: To: ;tag=819601190 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 4 [ 59]: Call-ID: 3f89999821936b5271d44c61398d948d@192.168.3.10:5060 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 6 [ 37]: Contact: [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 7 [115]: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 8 [ 39]: User-Agent: Yealink SIP-T20P 9.60.14.20 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 9 [ 51]: Allow-Events: talk,hold,conference,refer,check-sync [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [2012-12-26 13:51:53] VERBOSE[1871] chan_sip.c: --- (11 headers 0 lines) --- [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: = Looking for Call ID: 3f89999821936b5271d44c61398d948d@192.168.3.10:5060 (Checking To) --From tag as3fd12559 --To-tag 819601190 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3f89999821936b5271d44c61398d948d@192.168.3.10:5060' Request 102: Found [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: SIP response 180 to standard invite [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: build_route: Contact hop: [2012-12-26 13:51:53] VERBOSE[1871] chan_sip.c: list_route: hop: [2012-12-26 13:51:53] DEBUG[1842] devicestate.c: No provider found, checking channel drivers for SIP - 101 [2012-12-26 13:51:53] DEBUG[1842] chan_sip.c: Checking device state for peer 101 [2012-12-26 13:51:53] DEBUG[1842] devicestate.c: Changing state for SIP/101 - state 6 (Ringing) [2012-12-26 13:51:53] DEBUG[1842] devicestate.c: device 'SIP/101' state '6' [2012-12-26 13:51:53] DEBUG[1918] app_queue.c: Device 'SIP/101' changed to state '6' (Ringing) [2012-12-26 13:51:53] VERBOSE[9850] app_dial.c: -- SIP/101-00000069 is ringing [2012-12-26 13:51:53] DEBUG[9850] chan_dahdi.c: Requested indication 3 on channel DAHDI/1-1 [2012-12-26 13:51:53] DEBUG[9850] dsp.c: tone 1100, Ew=1.33E+04, Et=6.92E+07, s/n= 0.00 [2012-12-26 13:51:53] DEBUG[9850] res_rtp_asterisk.c: No remote address on RTP instance '0x223cce8' so dropping frame [2012-12-26 13:51:53] DEBUG[9850] dsp.c: tone 1100, Ew=6.98E+02, Et=6.27E+07, s/n= 0.00 [2012-12-26 13:51:53] DEBUG[9850] res_rtp_asterisk.c: No remote address on RTP instance '0x223cce8' so dropping frame [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Allocating new SIP dialog for 3b5d7c600c9e060f10dcfa950298155a@192.168.3.10:5060 - OPTIONS (No RTP) [2012-12-26 13:51:53] DEBUG[1871] acl.c: For destination '192.168.3.114', our source address is '192.168.3.10'. [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.3.10:5060 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Initializing initreq for method OPTIONS - callid 0e0837305866a1c90a4596900132482a@192.168.3.10:5060 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 0 [ 83]: OPTIONS sip:107@192.168.3.114:5060;rinstance=81baaa977341b065;transport=UDP SIP/2.0 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK6c5d7643 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as7bce5c0f [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 4 [ 73]: To: [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 5 [ 41]: Contact: [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 6 [ 59]: Call-ID: 0e0837305866a1c90a4596900132482a@192.168.3.10:5060 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 8 [ 22]: User-Agent: 0041762415 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 9 [ 35]: Date: Wed, 26 Dec 2012 09:51:53 GMT [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [2012-12-26 13:51:53] VERBOSE[1871] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.3.114:5060: OPTIONS sip:107@192.168.3.114:5060;rinstance=81baaa977341b065;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK6c5d7643 Max-Forwards: 70 From: "asterisk" ;tag=as7bce5c0f To: Contact: Call-ID: 0e0837305866a1c90a4596900132482a@192.168.3.10:5060 CSeq: 102 OPTIONS User-Agent: 0041762415 Date: Wed, 26 Dec 2012 09:51:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #36049 [2012-12-26 13:51:53] DEBUG[1871] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.3.114:5060 [2012-12-26 13:51:53] VERBOSE[1871] chan_sip.c: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK6c5d7643 Contact: To: ;tag=2d559063 From: "asterisk";tag=as7bce5c0f Call-ID: 0e0837305866a1c90a4596900132482a@192.168.3.10:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri User-Agent: Zoiper rev.11137 Allow-Events: presence, kpml Content-Length: 0 |