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SNR-7030 — звук в одну сторону Послк первого звонка IP-телефон не получает звук

Здравствуйте,

 

есть IP-телефон SNR-7030, подключён к Asterisk 1.8.16, астериск по Е1 связан с аналоговой станцией. После перезагрузки IP-телефона можно поговорить с аналоговым абонентом, но во время второго и последующих звонков в трубке IP-телефона тишина, звук с аналогового телефона не приходит. Аналоговый абонент при этом всё слышит. Если вместо IP-телефона использовать софтфон, то всё хорошо работает.

 

В чём дело? Что делать дальше, настраивать астериск или обращаться в техподдержку SNR?

 

Пытался сравнивать выхлоп астериска:

 

SNR

   ...
   -- Requested transfer capability: 0x00 - SPEECH
   -- Called DAHDI/g0/5396
   -- DAHDI/i1/5396-4c is proceeding passing it to SIP/7396-0000005a
   -- DAHDI/i1/5396-4c is ringing
   -- DAHDI/i1/5396-4c is making progress passing it to SIP/7396-0000005a
   -- DAHDI/i1/5396-4c answered SIP/7396-0000005a
   -- fixed jitterbuffer created on channel DAHDI/i1/5396-4c
   -- Span 1: Channel 0/1 got hangup request, cause 16
   -- Executing [h@macro-dialout-trunk:1] Macro("SIP/7396-0000005a", "hangupcall,") in new stack
   -- Executing [s@macro-hangupcall:1] GotoIf("SIP/7396-0000005a", "1?theend") in new stack
   -- Goto (macro-hangupcall,s,3)
   -- Executing [s@macro-hangupcall:3] Hangup("SIP/7396-0000005a", "") in new stack
 == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/7396-0000005a' in macro 'hangupcall'
 == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/7396-0000005a'
   -- Hungup 'DAHDI/i1/5396-4c'
   -- fixed jitterbuffer destroyed on channel DAHDI/i1/5396-4c
 == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/7396-0000005a' in macro 'dialout-trunk'
 == Spawn extension (from-internal, 5396, 7) exited non-zero on 'SIP/7396-0000005a'

 

софтфон (Blink):

   ...
   -- Requested transfer capability: 0x00 - SPEECH
   -- Called DAHDI/g0/5396
   -- DAHDI/i1/5396-4d is proceeding passing it to SIP/7397-0000005b
   -- DAHDI/i1/5396-4d is ringing
   -- DAHDI/i1/5396-4d is making progress passing it to SIP/7397-0000005b
   -- DAHDI/i1/5396-4d answered SIP/7397-0000005b
   -- fixed jitterbuffer created on channel DAHDI/i1/5396-4d
   -- Span 1: Channel 0/1 got hangup request, cause 16
   -- Executing [h@macro-dialout-trunk:1] Macro("SIP/7397-0000005b", "hangupcall,") in new stack
   -- Executing [s@macro-hangupcall:1] GotoIf("SIP/7397-0000005b", "1?theend") in new stack
   -- Goto (macro-hangupcall,s,3)
   -- Executing [s@macro-hangupcall:3] Hangup("SIP/7397-0000005b", "") in new stack
 == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/7397-0000005b' in macro 'hangupcall'
 == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/7397-0000005b'
   -- Hungup 'DAHDI/i1/5396-4d'
   -- fixed jitterbuffer destroyed on channel DAHDI/i1/5396-4d
 == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/7397-0000005b' in macro 'dialout-trunk'
 == Spawn extension (from-internal, 5396, 7) exited non-zero on 'SIP/7397-0000005b'

фатальных различий не увидел.

:(

Edited by muon

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нужно смотреть сип пакеты и есть ли rtp трафик. есть ли маршрутизатор с nat?

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NAT нет, LAN, абонент в одной подсети, астериск в другой.

IP у телефона с DHCP, не резервированный, может разные получать.

 

RTP ходит:

Sent RTP packet to      192.168.20.116:10036 (type 00, seq 036509, ts 082400, len 000160)
Got  RTP packet from    192.168.20.116:10036 (type 00, seq 022940, ts 1225231604, len 000160)
Sent RTP packet to      192.168.20.116:10036 (type 00, seq 036510, ts 082560, len 000160)
Got  RTP packet from    192.168.20.116:10036 (type 00, seq 022941, ts 1225231764, len 000160)
Sent RTP packet to      192.168.20.116:10036 (type 00, seq 036511, ts 082720, len 000160)
Got  RTP packet from    192.168.20.116:10036 (type 00, seq 022942, ts 1225231924, len 000160)
Sent RTP packet to      192.168.20.116:10036 (type 00, seq 036512, ts 082880, len 000160)
Got  RTP packet from    192.168.20.116:10036 (type 00, seq 022943, ts 1225232084, len 000160)

 

sip debug на проблемный номер, в простое:

<--- SIP read from UDP:192.168.20.116:5060 --->
REGISTER sip:asterisk0.msun.int SIP/2.0
Via: SIP/2.0/UDP 192.168.20.116:5060;branch=z9hG4bK19300236068938577;rport
From: Namen <sip:7396@asterisk0.msun.int:5060>;tag=300359317
To: Namen <sip:7396@asterisk0.msun.int:5060>
Call-ID: 1992342718936-119531866030069@192.168.20.116
CSeq: 61 REGISTER
Contact: <sip:7396@192.168.20.116:5060>
Max-Forwards: 70
Expires: 87
Supported: path
User-Agent: Greenlite ATOM V2.0
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.20.116:5060 (NAT)

<--- Transmitting (no NAT) to 192.168.20.116:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.20.116:5060;branch=z9hG4bK19300236068938577;received=192.168.20.116;rport=5060
From: Namen <sip:7396@asterisk0.msun.int:5060>;tag=300359317
To: Namen <sip:7396@asterisk0.msun.int:5060>;tag=as3fbfbcd5
Call-ID: 1992342718936-119531866030069@192.168.20.116
CSeq: 61 REGISTER
Server: FPBX-2.10.0(1.8.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b1bd3bd"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1992342718936-119531866030069@192.168.20.116' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.20.116:5060 --->
REGISTER sip:asterisk0.msun.int SIP/2.0
Via: SIP/2.0/UDP 192.168.20.116:5060;branch=z9hG4bK456630781636429961;rport
From: Namen <sip:7396@asterisk0.msun.int:5060>;tag=300359317
To: Namen <sip:7396@asterisk0.msun.int:5060>
Call-ID: 1992342718936-119531866030069@192.168.20.116
CSeq: 62 REGISTER
Contact: <sip:7396@192.168.20.116:5060>
Authorization: Digest username="7396", realm="asterisk", nonce="1b1bd3bd", uri="sip:asterisk0.msun.int", response="3fefc3f0163a787543ed5478e5dc8b04", algorithm=MD5
Max-Forwards: 70
Expires: 87
Supported: path
User-Agent: Greenlite ATOM V2.0
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.20.116:5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.20.116:5060:
OPTIONS sip:7396@192.168.20.116:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.79:5060;branch=z9hG4bK01a3e1fe
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.10.79>;tag=as754d0017
To: <sip:7396@192.168.20.116:5060>
Contact: <sip:Unknown@192.168.10.79:5060>
Call-ID: 74754af1257dfffc4711040a6e753e2a@192.168.10.79:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.0(1.8.16.0)
Date: Tue, 16 Oct 2012 04:07:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 192.168.20.116:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.20.116:5060;branch=z9hG4bK456630781636429961;received=192.168.20.116;rport=5060
From: Namen <sip:7396@asterisk0.msun.int:5060>;tag=300359317
To: Namen <sip:7396@asterisk0.msun.int:5060>;tag=as3fbfbcd5
Call-ID: 1992342718936-119531866030069@192.168.20.116
CSeq: 62 REGISTER
Server: FPBX-2.10.0(1.8.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 87
Contact: <sip:7396@192.168.20.116:5060>;expires=87
Date: Tue, 16 Oct 2012 04:07:07 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '45f79b775055328d2ee0b01a0bd74816@192.168.10.79:5060' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.20.116:5060:
NOTIFY sip:7396@192.168.20.116:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.79:5060;branch=z9hG4bK61403ec8
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.10.79>;tag=as37819252
To: <sip:7396@192.168.20.116:5060>
Contact: <sip:Unknown@192.168.10.79:5060>
Call-ID: 45f79b775055328d2ee0b01a0bd74816@192.168.10.79:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-2.10.0(1.8.16.0)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*97@192.168.10.79
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of SIP dialog '1992342718936-119531866030069@192.168.20.116' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.20.116:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.79:5060;branch=z9hG4bK01a3e1fe
From: "Unknown" <sip:Unknown@192.168.10.79>;tag=as754d0017
To: <sip:7396@192.168.20.116:5060>
Call-ID: 74754af1257dfffc4711040a6e753e2a@192.168.10.79:5060
CSeq: 102 OPTIONS
Contact: <sip:7396@192.168.20.116:5060>
Supported: 100rel, replaces, timer
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Accept: application/sdp, message/sipfrag, application/dtmf-relay
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '74754af1257dfffc4711040a6e753e2a@192.168.10.79:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.20.116:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.79:5060;branch=z9hG4bK61403ec8
From: "Unknown" <sip:Unknown@192.168.10.79>;tag=as37819252
To: <sip:7396@192.168.20.116:5060>
Call-ID: 45f79b775055328d2ee0b01a0bd74816@192.168.10.79:5060
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '45f79b775055328d2ee0b01a0bd74816@192.168.10.79:5060' Method: NOTIFY
Really destroying SIP dialog '1992342718936-119531866030069@192.168.20.116' Method: REGISTER

Мне это не нравится, слишком много всего, ещё и ответ 401. Blink (софтфон), например, постоянно посылает только

<--- SIP read from UDP:192.168.20.12:5060 --->
jaK
<------------->

и изредка OPTIONS, на который получает ответ 200.

 

Посмотрю ещё во время звонка.

Edited by muon

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