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Asterisk + SPA3102 проблема с PSTN Disconnect Tone

Вопросы по использованию и настройке IP телефонов, шлюзов и всего прочего

Модераторы: april22, Zavr2008

Asterisk + SPA3102 проблема с PSTN Disconnect Tone

Сообщение vitas » 17 ноя 2016, 00:49

Приветствую вас, коллеги
Вот уже несколько дней не могу разобраться с вот такой проблемой.
Имеется такое устройство как Linksys SPA3102, к нему подключен аналоговый телефон (в порт FXS - Phone), линия PSTN (в порт FXO - Line), в порт internet воткнут кабель который смотрит в локальную сеть 192.168.2.0/24
SPA3102 соиденен с Asterisk через trunk SIP. Входящая связь работает стабильно по такому сценарию: когда звонят на линию PSTN (порт FXO) вызов перебрасывается на Asterisk и попадает в меню IVR.
Когда звоню аналоговым телефоном который воткнут в порт FXS - связь не стабильная. То-есть:
1. звоню с FXS на GSM мобильник , после разговора если FXS бросает трубку первым срабатывает "VoIP Call Ended"
2. звоню с FXS на GSM мобильник , после разговора если GSM бросает трубку первым срабатывает "PSTN Disconnect Tone" (вот здесь и возникает проблема)
- после того как GSM бросил трубку первым и сработал "PSTN Disconnect Tone" . Если FXS снова позвонит на GSM или другой внешний номер, то происходит один гудок - GSM звонит один раз и FXS срывает вызов вот с таким кодом "Last PSTN Disconnect Reason: PSTN Disconnect Tone" . Если сразу же после этого позвонить второй или третий раз с FXS на GSM соединение происходит нормально до тех пор пока FXS ложит трубку первым и с этим срабатывает "Last PSTN Disconnect Reason: VoIP Call Ended". Происходит то же самое когда звонят внешние номера на линию PSTN и бросают трубку первым - то-есть: PSTN Disconnect Tone => после этого FXS не может звонить на внешние номера с первого раза ( а лишь только со второго или третьего раза). Думал проблема в телефоне который воткнут в FXS, пробовал звонить с софтфона происходить тоже самое.

При отключении функции "PSTN Disconnect Tone" = no линксис не сбрасывает линию если трубку ложит первый FXS (что проблиматично в случаее если звонок был произведен на IVR который бесконечно говорит)

в логах ничего странного не заметил

Может кто встречался с такой проблемой ?

--Настройки линксис в html файле-- http://dl.free.fr/vdjUwJefF

[Показать] Спойлер: Лог с линксис нормального исходящего вызова (с софтфона на GSM)
[Лог с линксис нормального исходящего вызова (с софтфона на GSM)]

Nov 16 19:56:41 192.168.2.22 [0]<<192.168.2.21: 5060(561)
Nov 16 19:56:41 192.168.2.22 [0]<<192.168.2.21: 5060(561)
Nov 16 19:56:41 OPTIONS sip: 6001@192.168.2.22:5060 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK5c056d32#015#012Max-Forwards: 70#015#012From: "asterisk" <sip:asterisk@192.168.2.21>;tag=as130cda40#015#012To: <sip:6001@192.168.2.22:5060>#015#012Contact: <sip:asterisk@192.168.2.21:5060>#015#012Call-ID: 032a888e0939cc7d6ccc8c00028e1bb6@192.168.2.21:5060#015#012CSeq: 102 OPTIONS#015#012User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1#015#012Date: Wed, 16 Nov 2016 18:56:41 GMT#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE#015#012Supported: replaces, timer#015#012Content-Length: 0#015#012#015
Nov 16 19:56:41 192.168.2.22
Nov 16 19:56:41 192.168.2.22
Nov 16 19:56:41 192.168.2.22 [0]->192.168.2.21: 5060(427)
Nov 16 19:56:41 192.168.2.22 [0]->192.168.2.21: 5060(427)
Nov 16 19:56:41 192.168.2.22 SIP/2.0 200 OK#015#012To: <sip:6001@192.168.2.22:5060>;tag=37dced9d79415fc2i0#015#012From: "asterisk" <sip:asterisk@192.168.2.21>;tag=as130cda40#015#012Call-ID: 032a888e0939cc7d6ccc8c00028e1bb6@192.168.2.21:5060#015#012CSeq: 102 OPTIONS#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK5c056d32#015#012Server: Linksys/SPA3102-5.2.13(GW002)#015#012Content-Length: 0#015#012Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER#015#012Supported: x-sipura, replaces#015#012#015
Nov 16 19:56:41 192.168.2.22
Nov 16 19:56:41 192.168.2.22
Nov 16 19:57:13 192.168.2.22 [1]<<192.168.2.21: 5060(789)
Nov 16 19:57:13 192.168.2.22 [1]<<192.168.2.21: 5060(789)
Nov 16 19:57:13 asterisk rsyslogd-2007: action 'action 17' suspended, next retry is Wed Nov 16 19:58:43 2016 [try http://www.rsyslog.com/e/2007 ]
Nov 16 19:57:13 INVITE sip: 0667XXXXXX@192.168.2.22 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK726491bd#015#012Max-Forwards: 70#015#012From: <sip:6003@192.168.2.21>;tag=as704d2d9c#015#012To: <sip:0667XXXXXX@192.168.2.22>#015#012Contact: <sip:6003@192.168.2.21:5060>#015#012Call-ID: 5fed1e0775ca8bfb679962bf1efe801a@192.168.2.21:5060#015#012CSeq: 102 INVITE#015#012User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1#015#012Date: Wed, 16 Nov 2016 18:57:13 GMT#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE#015#012Supported: replaces, timer#015#012Content-Type: application/sdp#015#012Content-Length: 214#015#012#015#012v=0#015#012o=root 1126008215 1126008215 IN IP4 192.168.2.21#015#012s=Asterisk PBX 11.13.1~dfsg-2+b1#015#012c=IN IP4 192.168.2.21#015#012t=0 0#015#012m=audio 19880 RTP/AVP 8 0#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=ptime:20#015#012a=sendrecv#015
Nov 16 19:57:13 192.168.2.22
Nov 16 19:57:13 192.168.2.22
Nov 16 19:57:13 192.168.2.22 [1]->192.168.2.21: 5060(299)
Nov 16 19:57:13 192.168.2.22 [1]->192.168.2.21: 5060(299)
Nov 16 19:57:13 192.168.2.22 SIP/2.0 100 Trying#015#012To: <sip:0667XXXXXX@192.168.2.22>#015#012From: <sip:6003@192.168.2.21>;tag=as704d2d9c#015#012Call-ID: 5fed1e0775ca8bfb679962bf1efe801a@192.168.2.21:5060#015#012CSeq: 102 INVITE#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK726491bd#015#012Server: Linksys/SPA3102-5.2.13(GW002)#015#012Content-Length: 0#015#012#015
Nov 16 19:57:13 192.168.2.22
Nov 16 19:57:13 192.168.2.22
Nov 16 19:57:13 192.168.2.22 [1: 0]AUD ALLOC CALL (port=17994)
Nov 16 19:57:13 192.168.2.22 [1: 0]RTP Rx Up
Nov 16 19:57:13 192.168.2.22 AUD: Stop PSTN Tone
Nov 16 19:57:13 192.168.2.22 [1]->192.168.2.21: 5060(811)
Nov 16 19:57:13 192.168.2.22 [1]->192.168.2.21: 5060(811)
Nov 16 19:57:13 192.168.2.22 SIP/2.0 200 OK#015#012To: <sip:0667XXXXXX@192.168.2.22>;tag=6e867a2e766820bfi1#015#012From: <sip:6003@192.168.2.21>;tag=as704d2d9c#015#012Call-ID: 5fed1e0775ca8bfb679962bf1efe801a@192.168.2.21:5060#015#012CSeq: 102 INVITE#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK726491bd#015#012Contact: <sip:0667XXXXXX@192.168.2.22:5061>#015#012Server: Linksys/SPA3102-5.2.13(GW002)#015#012Remote-Party-ID: <sip:6002@192.168.2.21>;screen=yes;party=called#015#012Content-Length: 255#015#012Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER#015#012Supported: x-sipura, replaces#015#012Content-Type: application/sdp#015#012#015#012v=0#015#012o=- 7459478 7459478 IN IP4 192.168.2.22#015#012s=-#015#012c=IN IP4 192.168.2.22#015#012t=0 0#015#012m=audio 17994 RTP/AVP 8 100 101#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:100 NSE/8000#015#012a=fmtp:100 192-193#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=ptime:20#015#012a=sendrecv#015
Nov 16 19:57:13 192.168.2.22
Nov 16 19:57:13 192.168.2.22
Nov 16 19:57:13 192.168.2.22 [1]<<192.168.2.21: 5060(407)
Nov 16 19:57:13 192.168.2.22 [1]<<192.168.2.21: 5060(407)
Nov 16 19:57:13 ACK sip: 0667XXXXXX@192.168.2.22:5061 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK3a6b7f7f#015#012Max-Forwards: 70#015#012From: <sip:6003@192.168.2.21>;tag=as704d2d9c#015#012To: <sip:0667XXXXXX@192.168.2.22>;tag=6e867a2e766820bfi1#015#012Contact: <sip:6003@192.168.2.21:5060>#015#012Call-ID: 5fed1e0775ca8bfb679962bf1efe801a@192.168.2.21:5060#015#012CSeq: 102 ACK#015#012User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1#015#012Content-Length: 0#015#012#015
Nov 16 19:57:13 192.168.2.22
Nov 16 19:57:13 192.168.2.22
Nov 16 19:57:13 192.168.2.22 CC: Connected
Nov 16 19:57:13 192.168.2.22 AUD: Stop PSTN Tone
Nov 16 19:57:13 192.168.2.22 [1: 0]ENC INIT 8
Nov 16 19:57:13 192.168.2.22 [1: 0]RTP Tx Up (pt=8->c0a80215:19880)
Nov 16 19:57:13 192.168.2.22 [1: 0]RTCP Tx Up
Nov 16 19:57:13 192.168.2.22 [1: 0]RTP Rx 1st PKT @17994(2)
Nov 16 19:57:13 192.168.2.22 [1]adp line session start
Nov 16 19:57:13 192.168.2.22 [1]adp line session start
Nov 16 19:57:14 192.168.2.22 [1: 0]DEC INIT 8
Nov 16 19:57:28 192.168.2.22 [1]<<192.168.2.21: 5060(561)
Nov 16 19:57:28 192.168.2.22 [1]<<192.168.2.21: 5060(561)
Nov 16 19:57:28 OPTIONS sip: 6002@192.168.2.22:5061 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK257b9ef5#015#012Max-Forwards: 70#015#012From: "asterisk" <sip:asterisk@192.168.2.21>;tag=as7578bf18#015#012To: <sip:6002@192.168.2.22:5061>#015#012Contact: <sip:asterisk@192.168.2.21:5060>#015#012Call-ID: 5748c9f82d9c6876620ae1586827578d@192.168.2.21:5060#015#012CSeq: 102 OPTIONS#015#012User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1#015#012Date: Wed, 16 Nov 2016 18:57:28 GMT#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE#015#012Supported: replaces, timer#015#012Content-Length: 0#015#012#015
Nov 16 19:57:28 192.168.2.22
Nov 16 19:57:28 192.168.2.22
Nov 16 19:57:28 192.168.2.22 [1]->192.168.2.21: 5060(426)
Nov 16 19:57:28 192.168.2.22 [1]->192.168.2.21: 5060(426)
Nov 16 19:57:28 192.168.2.22 SIP/2.0 200 OK#015#012To: <sip:6002@192.168.2.22:5061>;tag=13a46c5b22b98bai1#015#012From: "asterisk" <sip:asterisk@192.168.2.21>;tag=as7578bf18#015#012Call-ID: 5748c9f82d9c6876620ae1586827578d@192.168.2.21:5060#015#012CSeq: 102 OPTIONS#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK257b9ef5#015#012Server: Linksys/SPA3102-5.2.13(GW002)#015#012Content-Length: 0#015#012Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER#015#012Supported: x-sipura, replaces#015#012#015
Nov 16 19:57:28 192.168.2.22
Nov 16 19:57:28 192.168.2.22
Nov 16 19:57:29 192.168.2.22 [1: 0]LAT-- 6(2)
Nov 16 19:57:37 192.168.2.22 [0]->192.168.2.21: 5060(621)
Nov 16 19:57:37 192.168.2.22 [0]->192.168.2.21: 5060(621)
Nov 16 19:57:37 REGISTER sip: 192.168.2.21 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.2.22:5060;branch=z9hG4bK-51d0e73b#015#012From: <sip:6001@192.168.2.21>;tag=f8a48be1b882b64eo0#015#012To: <sip:6001@192.168.2.21>#015#012Call-ID: f71886a7-7dc68421@192.168.2.22#015#012CSeq: 6428 REGISTER#015#012Max-Forwards: 70#015#012Authorization: Digest username="6001",realm="asterisk",nonce="40004a2d",uri="sip:192.168.2.21",algorithm=MD5,response="335b933d5bab4daf00ab38839a705bcc"#015#012Contact: <sip:6001@192.168.2.22:5060>;expires=300#015#012User-Agent: Linksys/SPA3102-5.2.13(GW002)#015#012Content-Length: 0#015#012Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER#015#012Supported: x-sipura, replaces#015#012#015
Nov 16 19:57:37 192.168.2.22
Nov 16 19:57:37 192.168.2.22
Nov 16 19:57:37 192.168.2.22 [0]<<192.168.2.21: 5060(525)
Nov 16 19:57:37 192.168.2.22 [0]<<192.168.2.21: 5060(525)
Nov 16 19:57:37 192.168.2.22 SIP/2.0 401 Unauthorized#015#012Via: SIP/2.0/UDP 192.168.2.22:5060;branch=z9hG4bK-51d0e73b;received=192.168.2.22#015#012From: <sip:6001@192.168.2.21>;tag=f8a48be1b882b64eo0#015#012To: <sip:6001@192.168.2.21>;tag=as05ff69e2#015#012Call-ID: f71886a7-7dc68421@192.168.2.22#015#012CSeq: 6428 REGISTER#015#012Server: Asterisk PBX 11.13.1~dfsg-2+b1#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE#015#012Supported: replaces, timer#015#012WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7c20d68e"#015#012Content-Length: 0#015#012#015
Nov 16 19:57:37 192.168.2.22
Nov 16 19:57:37 192.168.2.22
Nov 16 19:57:37 192.168.2.22 [0]->192.168.2.21: 5060(621)
Nov 16 19:57:37 192.168.2.22 [0]->192.168.2.21: 5060(621)
Nov 16 19:57:37 REGISTER sip: 192.168.2.21 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.2.22:5060;branch=z9hG4bK-18e55c5c#015#012From: <sip:6001@192.168.2.21>;tag=f8a48be1b882b64eo0#015#012To: <sip:6001@192.168.2.21>#015#012Call-ID: f71886a7-7dc68421@192.168.2.22#015#012CSeq: 6429 REGISTER#015#012Max-Forwards: 70#015#012Authorization: Digest username="6001",realm="asterisk",nonce="7c20d68e",uri="sip:192.168.2.21",algorithm=MD5,response="5c58f0974eeb1e2d26c2bb800c0653fa"#015#012Contact: <sip:6001@192.168.2.22:5060>;expires=300#015#012User-Agent: Linksys/SPA3102-5.2.13(GW002)#015#012Content-Length: 0#015#012Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER#015#012Supported: x-sipura, replaces#015#012#015
Nov 16 19:57:37 192.168.2.22
Nov 16 19:57:37 192.168.2.22
Nov 16 19:57:37 192.168.2.22 [0]<<192.168.2.21: 5060(561)
Nov 16 19:57:37 192.168.2.22 [0]<<192.168.2.21: 5060(561)
Nov 16 19:57:37 OPTIONS sip: 6001@192.168.2.22:5060 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK6fb17502#015#012Max-Forwards: 70#015#012From: "asterisk" <sip:asterisk@192.168.2.21>;tag=as74bedc3d#015#012To: <sip:6001@192.168.2.22:5060>#015#012Contact: <sip:asterisk@192.168.2.21:5060>#015#012Call-ID: 68f098384cfc0227174323ea1bd4cdc9@192.168.2.21:5060#015#012CSeq: 102 OPTIONS#015#012User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1#015#012Date: Wed, 16 Nov 2016 18:57:37 GMT#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE#015#012Supported: replaces, timer#015#012Content-Length: 0#015#012#015
Nov 16 19:57:37 192.168.2.22
Nov 16 19:57:37 192.168.2.22
Nov 16 19:57:37 192.168.2.22 [0]->192.168.2.21: 5060(427)
Nov 16 19:57:37 192.168.2.22 [0]->192.168.2.21: 5060(427)
Nov 16 19:57:37 192.168.2.22 SIP/2.0 200 OK#015#012To: <sip:6001@192.168.2.22:5060>;tag=37dced9d79415fc2i0#015#012From: "asterisk" <sip:asterisk@192.168.2.21>;tag=as74bedc3d#015#012Call-ID: 68f098384cfc0227174323ea1bd4cdc9@192.168.2.21:5060#015#012CSeq: 102 OPTIONS#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK6fb17502#015#012Server: Linksys/SPA3102-5.2.13(GW002)#015#012Content-Length: 0#015#012Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER#015#012Supported: x-sipura, replaces#015#012#015
Nov 16 19:57:37 192.168.2.22
Nov 16 19:57:37 192.168.2.22
Nov 16 19:57:37 192.168.2.22 [0]<<192.168.2.21: 5060(541)
Nov 16 19:57:37 192.168.2.22 [0]<<192.168.2.21: 5060(541)
Nov 16 19:57:37 192.168.2.22 SIP/2.0 200 OK#015#012Via: SIP/2.0/UDP 192.168.2.22:5060;branch=z9hG4bK-18e55c5c;received=192.168.2.22#015#012From: <sip:6001@192.168.2.21>;tag=f8a48be1b882b64eo0#015#012To: <sip:6001@192.168.2.21>;tag=as05ff69e2#015#012Call-ID: f71886a7-7dc68421@192.168.2.22#015#012CSeq: 6429 REGISTER#015#012Server: Asterisk PBX 11.13.1~dfsg-2+b1#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE#015#012Supported: replaces, timer#015#012Expires: 300#015#012Contact: <sip:6001@192.168.2.22:5060>;expires=300#015#012Date: Wed, 16 Nov 2016 18:57:37 GMT#015#012Content-Length: 0#015#012#015
Nov 16 19:57:37 192.168.2.22
Nov 16 19:57:37 192.168.2.22
Nov 16 19:57:37 192.168.2.22 [1: 0]LAT-- 6(2)
Nov 16 19:57:45 192.168.2.22 [1: 0]LAT-- 5(2)
Nov 16 19:57:46 192.168.2.22 FXO: PSTN Disconnect Tone
Nov 16 19:57:46 192.168.2.22 AUD: Stop PSTN Tone
Nov 16 19:57:46 192.168.2.22 AUD: Stop PSTN Tone
Nov 16 19:57:46 192.168.2.22 [1: 0]AUD Rel Call
Nov 16 19:57:46 192.168.2.22 [1]->192.168.2.21: 5060(362)
Nov 16 19:57:46 192.168.2.22 [1]->192.168.2.21: 5060(362)
Nov 16 19:57:46 BYE sip: 6003@192.168.2.21:5060 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.2.22:5061;branch=z9hG4bK-c49c0143#015#012From: <sip:0667XXXXXX@192.168.2.22>;tag=6e867a2e766820bfi1#015#012To: <sip:6003@192.168.2.21>;tag=as704d2d9c#015#012Call-ID: 5fed1e0775ca8bfb679962bf1efe801a@192.168.2.21:5060#015#012CSeq: 101 BYE#015#012Max-Forwards: 70#015#012User-Agent: Linksys/SPA3102-5.2.13(GW002)#015#012Content-Length: 0#015#012#015
Nov 16 19:57:46 192.168.2.22
Nov 16 19:57:46 192.168.2.22
Nov 16 19:57:46 asterisk asterisk[4971]: rc_avpair_new: unknown attribute 1490026597
Nov 16 19:57:46 192.168.2.22 [1]adp line session stop
Nov 16 19:57:46 192.168.2.22 [1]adp line session stop
Nov 16 19:57:46 --------------------------
Nov 16 19:57:46 --------------------------
Nov 16 19:57:46 192.168.2.22 [1] duration:33 s
Nov 16 19:57:46 192.168.2.22 [1] duration:33 s
Nov 16 19:57:46 192.168.2.22 path: nb_in, did:60, start at 3320 s
Nov 16 19:57:46 192.168.2.22 path: nb_in, did:60, start at 3320 s
Nov 16 19:57:46 192.168.2.22 path: nb_out, did:61, start at 3320 s
Nov 16 19:57:46 192.168.2.22 path: nb_out, did:61, start at 3320 s
Nov 16 19:57:46 192.168.2.22 path: full, did:62, start at 3320 s
Nov 16 19:57:46 192.168.2.22 path: full, did:62, start at 3320 s
Nov 16 19:57:46 --------------------------
Nov 16 19:57:46 --------------------------
Nov 16 19:57:46 192.168.2.22 [1]<<192.168.2.21: 5060(459)
Nov 16 19:57:46 192.168.2.22 [1]<<192.168.2.21: 5060(459)
Nov 16 19:57:46 192.168.2.22 SIP/2.0 200 OK#015#012Via: SIP/2.0/UDP 192.168.2.22:5061;branch=z9hG4bK-c49c0143;received=192.168.2.22#015#012From: <sip:0667XXXXXX@192.168.2.22>;tag=6e867a2e766820bfi1#015#012To: <sip:6003@192.168.2.21>;tag=as704d2d9c#015#012Call-ID: 5fed1e0775ca8bfb679962bf1efe801a@192.168.2.21:5060#015#012CSeq: 101 BYE#015#012Server: Asterisk PBX 11.13.1~dfsg-2+b1#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE#015#012Supported: replaces, timer#015#012Content-Length: 0#015#012#015
Nov 16 19:57:46 192.168.2.22
Nov 16 19:57:46 192.168.2.22
Nov 16 19:57:46 DLG Terminated 2e11e0
Nov 16 19:57:46 Sess Terminated
Nov 16 19:57:46 192.168.2.22 FXO: PSTN Disconnect Tone
Nov 16 19:57:46 192.168.2.22 AUD: Stop PSTN Tone
Nov 16 19:57:46 192.168.2.22 AUD: Stop PSTN Tone
^C


[Показать] Спойлер: Лог Asterisk
Connected to Asterisk 11.13.1~dfsg-2+b1 currently running on asterisk (pid = 4971)
== Using SIP RTP CoS mark 5
-- Executing [0667XXXXXX@internal-phones:1] Dial("SIP/6003-00000006", "SIP/6002/0667XXXXXX") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/6002/0667XXXXXX
-- SIP/6002-00000007 answered SIP/6003-00000006
> 0x7661fdd8 -- Probation passed - setting RTP source address to 192.168.2.12:4002
> 0x76520198 -- Probation passed - setting RTP source address to 192.168.2.22:17994
> 0x7661fdd8 -- Probation passed - setting RTP source address to 192.168.2.12:4002
== Spawn extension (internal-phones, 0667XXXXXX, 1) exited non-zero on 'SIP/6003-00000006'


[Показать] Спойлер: Лог с линксис сорваного исходящего вызова (с софтфона на GSM) = > тоже самое происходит с FXS
root@asterisk:~# tail -f /var/log/syslog
Nov 16 20:14:46 192.168.2.22 [1]<<192.168.2.21: 5060(561)
Nov 16 20:14:46 192.168.2.22 [1]<<192.168.2.21: 5060(561)
Nov 16 20:14:46 OPTIONS sip: 6002@192.168.2.22:5061 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK52b36278#015#012Max-Forwards: 70#015#012From: "asterisk" <sip:asterisk@192.168.2.21>;tag=as3485e105#015#012To: <sip:6002@192.168.2.22:5061>#015#012Contact: <sip:asterisk@192.168.2.21:5060>#015#012Call-ID: 1ca0d6434be64c37348f292b1a865787@192.168.2.21:5060#015#012CSeq: 102 OPTIONS#015#012User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1#015#012Date: Wed, 16 Nov 2016 19:14:46 GMT#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE#015#012Supported: replaces, timer#015#012Content-Length: 0#015#012#015
Nov 16 20:14:46 192.168.2.22
Nov 16 20:14:46 192.168.2.22
Nov 16 20:14:46 192.168.2.22 [1]->192.168.2.21: 5060(426)
Nov 16 20:14:46 192.168.2.22 [1]->192.168.2.21: 5060(426)
Nov 16 20:14:46 192.168.2.22 SIP/2.0 200 OK#015#012To: <sip:6002@192.168.2.22:5061>;tag=13a46c5b22b98bai1#015#012From: "asterisk" <sip:asterisk@192.168.2.21>;tag=as3485e105#015#012Call-ID: 1ca0d6434be64c37348f292b1a865787@192.168.2.21:5060#015#012CSeq: 102 OPTIONS#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK52b36278#015#012Server: Linksys/SPA3102-5.2.13(GW002)#015#012Content-Length: 0#015#012Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER#015#012Supported: x-sipura, replaces#015#012#015
Nov 16 20:14:46 192.168.2.22
Nov 16 20:14:46 192.168.2.22
Nov 16 20:15:00 192.168.2.22 [1]<<192.168.2.21: 5060(799)
Nov 16 20:15:00 192.168.2.22 [1]<<192.168.2.21: 5060(799)
Nov 16 20:15:00 INVITE sip: 0667XXXXXX@192.168.2.22:5061 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK29c2b891#015#012Max-Forwards: 70#015#012From: <sip:6003@192.168.2.21>;tag=as6bec102e#015#012To: <sip:0667XXXXXX@192.168.2.22:5061>#015#012Contact: <sip:6003@192.168.2.21:5060>#015#012Call-ID: 71bc996608ff74e13d7b76997d876c7f@192.168.2.21:5060#015#012CSeq: 102 INVITE#015#012User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1#015#012Date: Wed, 16 Nov 2016 19:15:00 GMT#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE#015#012Supported: replaces, timer#015#012Content-Type: application/sdp#015#012Content-Length: 214#015#012#015#012v=0#015#012o=root 1530759495 1530759495 IN IP4 192.168.2.21#015#012s=Asterisk PBX 11.13.1~dfsg-2+b1#015#012c=IN IP4 192.168.2.21#015#012t=0 0#015#012m=audio 14604 RTP/AVP 8 0#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=ptime:20#015#012a=sendrecv#015
Nov 16 20:15:00 192.168.2.22
Nov 16 20:15:00 192.168.2.22
Nov 16 20:15:00 192.168.2.22 [1]->192.168.2.21: 5060(304)
Nov 16 20:15:00 192.168.2.22 [1]->192.168.2.21: 5060(304)
Nov 16 20:15:00 192.168.2.22 SIP/2.0 100 Trying#015#012To: <sip:0667XXXXXX@192.168.2.22:5061>#015#012From: <sip:6003@192.168.2.21>;tag=as6bec102e#015#012Call-ID: 71bc996608ff74e13d7b76997d876c7f@192.168.2.21:5060#015#012CSeq: 102 INVITE#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK29c2b891#015#012Server: Linksys/SPA3102-5.2.13(GW002)#015#012Content-Length: 0#015#012#015
Nov 16 20:15:00 192.168.2.22
Nov 16 20:15:00 192.168.2.22
Nov 16 20:15:00 192.168.2.22 [1: 0]AUD ALLOC CALL (port=18008)
Nov 16 20:15:00 192.168.2.22 [1: 0]RTP Rx Up
Nov 16 20:15:00 192.168.2.22 AUD: Stop PSTN Tone
Nov 16 20:15:00 192.168.2.22 [1]->192.168.2.21: 5060(816)
Nov 16 20:15:00 192.168.2.22 [1]->192.168.2.21: 5060(816)
Nov 16 20:15:00 192.168.2.22 SIP/2.0 200 OK#015#012To: <sip:0667XXXXXX@192.168.2.22:5061>;tag=b088721288cb8ae4i1#015#012From: <sip:6003@192.168.2.21>;tag=as6bec102e#015#012Call-ID: 71bc996608ff74e13d7b76997d876c7f@192.168.2.21:5060#015#012CSeq: 102 INVITE#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK29c2b891#015#012Contact: <sip:0667XXXXXX@192.168.2.22:5061>#015#012Server: Linksys/SPA3102-5.2.13(GW002)#015#012Remote-Party-ID: <sip:6002@192.168.2.21>;screen=yes;party=called#015#012Content-Length: 255#015#012Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER#015#012Supported: x-sipura, replaces#015#012Content-Type: application/sdp#015#012#015#012v=0#015#012o=- 7566159 7566159 IN IP4 192.168.2.22#015#012s=-#015#012c=IN IP4 192.168.2.22#015#012t=0 0#015#012m=audio 18008 RTP/AVP 8 100 101#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:100 NSE/8000#015#012a=fmtp:100 192-193#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=ptime:20#015#012a=sendrecv#015
Nov 16 20:15:00 192.168.2.22
Nov 16 20:15:00 192.168.2.22
Nov 16 20:15:00 192.168.2.22 [1]<<192.168.2.21: 5060(412)
Nov 16 20:15:00 192.168.2.22 [1]<<192.168.2.21: 5060(412)
Nov 16 20:15:00 ACK sip: 0667XXXXXX@192.168.2.22:5061 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.2.21:5060;branch=z9hG4bK60f5868a#015#012Max-Forwards: 70#015#012From: <sip:6003@192.168.2.21>;tag=as6bec102e#015#012To: <sip:0667XXXXXX@192.168.2.22:5061>;tag=b088721288cb8ae4i1#015#012Contact: <sip:6003@192.168.2.21:5060>#015#012Call-ID: 71bc996608ff74e13d7b76997d876c7f@192.168.2.21:5060#015#012CSeq: 102 ACK#015#012User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1#015#012Content-Length: 0#015#012#015
Nov 16 20:15:00 192.168.2.22
Nov 16 20:15:00 192.168.2.22
Nov 16 20:15:00 192.168.2.22 CC: Connected
Nov 16 20:15:00 192.168.2.22 AUD: Stop PSTN Tone
Nov 16 20:15:00 192.168.2.22 [1: 0]ENC INIT 8
Nov 16 20:15:00 192.168.2.22 [1: 0]RTP Tx Up (pt=8->c0a80215:14604)
Nov 16 20:15:00 192.168.2.22 [1: 0]RTCP Tx Up
Nov 16 20:15:00 192.168.2.22 [1]adp line session start
Nov 16 20:15:00 192.168.2.22 [1]adp line session start
Nov 16 20:15:00 192.168.2.22 [1: 0]RTP Rx 1st PKT @18008(2)
Nov 16 20:15:00 192.168.2.22 [1: 0]DEC INIT 8
Nov 16 20:15:07 192.168.2.22 FXO: PSTN Disconnect Tone
Nov 16 20:15:07 192.168.2.22 AUD: Stop PSTN Tone
Nov 16 20:15:07 192.168.2.22 AUD: Stop PSTN Tone
Nov 16 20:15:07 192.168.2.22 [1: 0]AUD Rel Call
Nov 16 20:15:07 192.168.2.22 [1]->192.168.2.21: 5060(362)
Nov 16 20:15:07 192.168.2.22 [1]->192.168.2.21: 5060(362)
Nov 16 20:15:07 BYE sip: 6003@192.168.2.21:5060 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.2.22:5061;branch=z9hG4bK-d8b1f1ae#015#012From: <sip:0667XXXXXX@192.168.2.22>;tag=b088721288cb8ae4i1#015#012To: <sip:6003@192.168.2.21>;tag=as6bec102e#015#012Call-ID: 71bc996608ff74e13d7b76997d876c7f@192.168.2.21:5060#015#012CSeq: 101 BYE#015#012Max-Forwards: 70#015#012User-Agent: Linksys/SPA3102-5.2.13(GW002)#015#012Content-Length: 0#015#012#015
Nov 16 20:15:07 192.168.2.22
Nov 16 20:15:07 192.168.2.22
Nov 16 20:15:07 asterisk asterisk[4971]: rc_avpair_new: unknown attribute 1490026597
Nov 16 20:15:08 192.168.2.22 [1]adp line session stop
Nov 16 20:15:08 192.168.2.22 [1]adp line session stop
Nov 16 20:15:08 --------------------------
Nov 16 20:15:08 --------------------------
Nov 16 20:15:08 192.168.2.22 [1] duration:7 s
Nov 16 20:15:08 192.168.2.22 [1] duration:7 s
Nov 16 20:15:08 192.168.2.22 path: nb_in, did:60, start at 3414 s
Nov 16 20:15:08 192.168.2.22 path: nb_in, did:60, start at 3414 s
Nov 16 20:15:08 192.168.2.22 path: nb_out, did:61, start at 3414 s
Nov 16 20:15:08 192.168.2.22 path: nb_out, did:61, start at 3414 s
Nov 16 20:15:08 192.168.2.22 path: full, did:62, start at 3414 s
Nov 16 20:15:08 192.168.2.22 path: full, did:62, start at 3414 s
Nov 16 20:15:08 --------------------------
Nov 16 20:15:08 --------------------------
Nov 16 20:15:08 192.168.2.22 [1]<<192.168.2.21: 5060(459)
Nov 16 20:15:08 192.168.2.22 [1]<<192.168.2.21: 5060(459)
Nov 16 20:15:08 192.168.2.22 SIP/2.0 200 OK#015#012Via: SIP/2.0/UDP 192.168.2.22:5061;branch=z9hG4bK-d8b1f1ae;received=192.168.2.22#015#012From: <sip:0667XXXXXX@192.168.2.22>;tag=b088721288cb8ae4i1#015#012To: <sip:6003@192.168.2.21>;tag=as6bec102e#015#012Call-ID: 71bc996608ff74e13d7b76997d876c7f@192.168.2.21:5060#015#012CSeq: 101 BYE#015#012Server: Asterisk PBX 11.13.1~dfsg-2+b1#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE#015#012Supported: replaces, timer#015#012Content-Length: 0#015#012#015
Nov 16 20:15:08 192.168.2.22
Nov 16 20:15:08 192.168.2.22
Nov 16 20:15:08 DLG Terminated 2e0dd4
Nov 16 20:15:08 Sess Terminated
]


[Показать] Спойлер: Лог Asterisk сорваного исходящего вызова
asterisk*CLI>
== Using SIP RTP CoS mark 5
-- Executing [0667XXXXXX@internal-phones:1] Dial("SIP/6003-0000000d", "SIP/6002/0667XXXXXX") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/6002/0667XXXXXX
-- SIP/6002-0000000e answered SIP/6003-0000000d
> 0x76601ee0 -- Probation passed - setting RTP source address to 192.168.2.12:4008
> 0x7650e0d0 -- Probation passed - setting RTP source address to 192.168.2.22:18002
> 0x76601ee0 -- Probation passed - setting RTP source address to 192.168.2.12:4008
== Spawn extension (internal-phones, 0667XXXXXX, 1) exited non-zero on 'SIP/6003-0000000d'


[Показать] Спойлер: sip.conf
[6001]
type=friend
secret=password
username=6001
context=internal-phones
host=dynamic
disallow=all
allow=alaw
allow=ulaw
insecure=very
qualify=yes
canreinvite=no
dtmfmode=inband
nat=no


[6002]
username=6002
type=peer
secret=password
host=dynamic
disallow=all
allow=alaw
allow=ulaw
qualify=yes
insecure=very
canreinvite=no
dtmfmode=inband
nat=no

[6003]
type=friend
secret=password
context=internal-phones
host=dynamic
disallow=all
allow=alaw
allow=ulaw
dtmfmode=rfc2833
nat=no


[Показать] Спойлер: extensions.conf
[internal-phones]

exten => _6XXX,1,Dial(SIP/${EXTEN},60)
exten => _6XXX,n,Hangup

exten => _06XXXXXXXX,1,Dial(SIP/6002/${EXTEN})
06XXXXXXXX = Мобильные номера


[sip show peers]
asterisk*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
6001/6001 192.168.2.22 D No No 5060 OK (9 ms)
6002/6002 192.168.2.22 D No No 5061 OK (8 ms)
6003/6003 192.168.2.12 D No No 55359 OK (1 ms)

6001 = FXS Line 1
6002 = PSTN
6003 = Софтфон


сервер asterisk = 192.168.2.21
линксис = 192.168.2.22
софтфон = 192.168.2.12
vitas
 
Сообщений: 10
Зарегистрирован: 27 дек 2015, 02:31

Re: Asterisk + SPA3102 проблема с PSTN Disconnect Tone

Сообщение vitas » 17 ноя 2016, 05:52

ребята, проблема решилась
навела на мысль вот эта ссылка http://adminote.blogspot.fr/2009/08/spa ... ction.html
по видимому первый гудок вызова иногда был слишком короткий и засчитывался как отбойный гудок
значение Disconnect Tone: 440@-20,440@-20;1(0.5/0.5/1) поменял на 440@-20,440@-20;4(0.5/0.5/1) и все заработало отлично ;)
vitas
 
Сообщений: 10
Зарегистрирован: 27 дек 2015, 02:31


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