Проблемы с callback
Недавно начал настраивать callback на FreePBX 2.9.0.7. Настроил по след. схеме:
Inbound Routers (правило с приоритетом по ClID)-> Callback -> Disa.
При входящем звонке, система завершает разговор, и через 10 секунд (как указано в настройках Callback, а пробовал и через 0 сек.) перезванивает на позвонивший номер. А вот тут проблема. Система сразу же завершает исходящий звонок. Вот кусок лога
| Code: |
| [Sep 11 19:06:36] DEBUG[11035] chan_sip.c: Outgoing Call for 79268888888 [Sep 11 19:06:36] DEBUG[11035] chan_sip.c: Updating call counter for outgoing call [Sep 11 19:06:36] DEBUG[11035] chan_sip.c: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False Text flag: False [Sep 11 19:06:36] DEBUG[11035] chan_sip.c: ** Our prefcodec: 0x40 (slin) [Sep 11 19:06:36] DEBUG[11035] chan_sip.c: -- Done with adding codecs to SDP [Sep 11 19:06:36] DEBUG[11035] chan_sip.c: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Sep 11 19:06:36] DEBUG[11035] chan_sip.c: Initializing initreq for method INVITE - callid 3689a7dd101ab2fe5d0a4cd369dcc7f1@ss.callobok.ru [Sep 11 19:06:36] DEBUG[11035] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 188.120.245.137:5060 [Sep 11 19:06:36] VERBOSE[11035] app_dial.c: -- Called callobok_tmbrl_1/79268888888 [b][Sep 11 19:06:36] WARNING[11035] channel.c: No path to translate from SIP/callobok_tmbrl_1-000000c8 to Local/79268888888@from-internal-e7b6;2[/b] [Sep 11 19:06:36] DEBUG[11035] channel.c: Hanging up channel 'SIP/callobok_tmbrl_1-000000c8' [Sep 11 19:06:36] DEBUG[11035] chan_sip.c: Hangup call SIP/callobok_tmbrl_1-000000c8, SIP callid 3689a7dd101ab2fe5d0a4cd369dcc7f1@ss.callobok.ru [Sep 11 19:06:36] DEBUG[11035] chan_sip.c: Hanging up channel in state Down (not UP) [Sep 11 19:06:36] DEBUG[11035] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1c425808' [Sep 11 19:06:36] DEBUG[907] devicestate.c: No provider found, checking channel drivers for SIP - callobok_tmbrl_1 [Sep 11 19:06:36] DEBUG[907] chan_sip.c: Checking device state for peer callobok_tmbrl_1 [Sep 11 19:06:36] DEBUG[907] devicestate.c: Changing state for SIP/callobok_tmbrl_1 - state 1 (Not in use) [Sep 11 19:06:36] DEBUG[907] devicestate.c: device 'SIP/callobok_tmbrl_1' state '1' [Sep 11 19:06:36] DEBUG[933] app_queue.c: Device 'SIP/callobok_tmbrl_1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 11 19:06:36] DEBUG[11035] app_dial.c: Exiting with DIALSTATUS=CONGESTION. [Sep 11 19:06:36] DEBUG[11035] app_macro.c: Spawn extension (macro-dialout-trunk,s,20) exited non-zero on 'Local/79268888888@from-internal-e7b6;2' in macro 'dialout-trunk' [Sep 11 19:06:36] VERBOSE[11035] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'Local/79268888888@from-internal-e7b6;2' in macro 'dialout-trunk' [Sep 11 19:06:36] DEBUG[11035] pbx.c: Spawn extension (from-internal,79268888888,5) exited non-zero on 'Local/79268888888@from-internal-e7b6;2' [Sep 11 19:06:36] VERBOSE[11035] pbx.c: == Spawn extension (from-internal, 79268888888, 5) exited non-zero on 'Local/79268888888@from-internal-e7b6;2' [Sep 11 19:06:36] DEBUG[11035] channel.c: Soft-Hanging up channel 'Local/79268888888@from-internal-e7b6;2' [Sep 11 19:06:36] DEBUG[11035] pbx.c: Launching 'Hangup' |
что значит No path to translate from SIP/callobok_tmbrl_1-000000c8 to Local/79268888888@from-internal-e7b6;2 так в нете и не нашел. Пишут про несовместимость кодеков, но врядли. В настройках прописаны gsm&ulaw&alaw они поддерживаются оператором. Исходящие звонки идут без проблем. Работает webcallback через Asterisk API. А тут вот борода какая то. Буду благодарен если ткнете куда копать.
| Code: |
| [Sep 11 21:32:29] VERBOSE[16784] netsock.c: == Using UDPTL TOS bits 184 [Sep 11 21:32:29] VERBOSE[16784] netsock.c: == Using UDPTL CoS mark 5 [Sep 11 21:32:29] VERBOSE[16784] netsock2.c: == Using SIP RTP TOS bits 184 [Sep 11 21:32:29] VERBOSE[16784] netsock2.c: == Using SIP RTP CoS mark 5 [Sep 11 21:32:29] VERBOSE[16784] chan_sip.c: Audio is at 5061 [Sep 11 21:32:29] VERBOSE[16784] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Sep 11 21:32:29] VERBOSE[16784] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Sep 11 21:32:29] VERBOSE[16784] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Sep 11 21:32:29] VERBOSE[16784] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Sep 11 21:32:29] VERBOSE[16784] chan_sip.c: Reliably Transmitting (NAT) to 188.120.245.137:5060: INVITE sip:79268888888@ss.callobok.ru:5060 SIP/2.0 Via: SIP/2.0/UDP 178.XXX.XXX.XXX:5061;branch=z9hG4bK7f671452;rport Max-Forwards: 70 From: "2007059" ;tag=as0adc7750 To: Contact: Call-ID: 6a33b5926c7aadc7521d1c1727a1ab11@ss.callobok.ru CSeq: 102 INVITE User-Agent: FPBX-2.9.0(1.8.4.4) Date: Sun, 11 Sep 2011 17:32:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 v=0 o=root 1788014213 1788014213 IN IP4 178.XXX.XXX.XXX s=Asterisk PBX 1.8.4.4 c=IN IP4 178.XXX.XXX.XXX t=0 0 m=audio 19210 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Sep 11 21:32:29] VERBOSE[16784] app_dial.c: -- Called callobok_tmbrl_1/79268888888 [Sep 11 21:32:29] WARNING[16784] channel.c: No path to translate from SIP/callobok_tmbrl_1-000000e0 to Local/79268888888@from-internal-847a;2 [Sep 11 21:32:29] VERBOSE[16784] chan_sip.c: Scheduling destruction of SIP dialog '6a33b5926c7aadc7521d1c1727a1ab11@ss.callobok.ru' in 32000 ms (Method: INVITE) [Sep 11 21:32:29] VERBOSE[16784] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'Local/79268888888@from-internal-847a;2' in macro 'dialout-trunk' [Sep 11 21:32:29] VERBOSE[16784] pbx.c: == Spawn extension (from-internal, 79268888888, 5) exited non-zero on 'Local/79268888888@from-internal-847a;2' [Sep 11 21:32:29] VERBOSE[16784] pbx.c: -- Executing [h@from-internal:1] Hangup("Local/79268888888@from-internal-847a;2", "") in new stack [Sep 11 21:32:29] VERBOSE[16784] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'Local/79268888888@from-internal-847a;2' [Sep 11 21:32:29] VERBOSE[16780] manager.c: == Manager 'admin' logged off from 127.0.0.1 [Sep 11 21:32:30] VERBOSE[927] chan_sip.c: Retransmitting #1 (NAT) to 188.120.245.137:5060: CANCEL sip:79268888888@ss.callobok.ru:5060 SIP/2.0 Via: SIP/2.0/UDP 178.XXX.XXX.XXX:5061;branch=z9hG4bK3ef8fb1c;rport Max-Forwards: 70 From: "2007059" ;tag=as5aa66855 To: Call-ID: 4151b07d7a4ec6e872317b4135e5cf77@ss.callobok.ru CSeq: 103 CANCEL User-Agent: FPBX-2.9.0(1.8.4.4) Content-Length: 0 --- [Sep 11 21:32:30] VERBOSE[953] app.c: -- Message check requested for mailbox 298@device/folder INBOX but voicemail not loaded. [Sep 11 21:32:30] VERBOSE[953] app.c: -- Message check requested for mailbox 218@device/folder INBOX but voicemail not loaded. [Sep 11 21:32:30] VERBOSE[927] chan_sip.c: SIP/2.0 487 Request Terminated v: SIP/2.0/UDP 178.XXX.XXX.XXX:5061;branch=z9hG4bK3ef8fb1c;received=178.XXX.XXX.XXX;rport=62967 f: "2007059" ;tag=as5aa66855 t: ;tag=as2bf28e02 i: 4151b07d7a4ec6e872317b4135e5cf77@ss.callobok.ru CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO k: replaces, timer l: 0 [Sep 11 21:32:30] VERBOSE[927] chan_sip.c: --- (10 headers 0 lines) --- [Sep 11 21:32:30] VERBOSE[927] chan_sip.c: Transmitting (NAT) to 188.120.245.137:5060: ACK sip:79268888888@ss.callobok.ru:5060 SIP/2.0 Via: SIP/2.0/UDP 178.XXX.XXX.XXX:5061;branch=z9hG4bK3ef8fb1c;rport Max-Forwards: 70 From: "2007059" ;tag=as5aa66855 To: ;tag=as2bf28e02 Contact: Call-ID: 4151b07d7a4ec6e872317b4135e5cf77@ss.callobok.ru CSeq: 103 ACK User-Agent: FPBX-2.9.0(1.8.4.4) Content-Length: 0 --- [Sep 11 21:32:30] VERBOSE[927] chan_sip.c: SIP/2.0 200 OK v: SIP/2.0/UDP 178.XXX.XXX.XXX:5061;branch=z9hG4bK3ef8fb1c;received=178.XXX.XXX.XXX;rport=62967 f: "2007059" ;tag=as5aa66855 t: ;tag=as2bf28e02 i: 4151b07d7a4ec6e872317b4135e5cf77@ss.callobok.ru CSeq: 103 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO k: replaces, timer l: 0 |
| Quote: |
| Кодеки: g.711, g723 или g.729 (предпочтительнее) |
http://www.iamroot.ru/2009/04/%d0%b0%d1% ... d1%8f.html
| zlat wrote: | ||
| с сайта колобка |
| Quote: |
| Кодеки: g.711, g723 или g.729 (предпочтительнее) |
http://www.iamroot.ru/2009/04/%d0%b0%d1% ... d1%8f.html
Помогло! Извиняюсь что сразу не ответил. Но вот не пойму честно, почему. У меня до этого стояли кодеки ulaw&alaw&gsm и ведь шли звонки. И обычные и по вебкаллбеку. А по обычному почему отказывались?
Большое спасибо за помощь!