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Голосовые сообщения об ошибке не приходят с транков

https://forum.asterisk.ru/viewtopic.php?t=5025

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Голосовые сообщения об ошибке не приходят с транков

Добавлено: 10 дек 2014, 23:38
gaspaarr
Собственно возник вопрос, что с внешних транков не приходят сообщения о недоступности абонента (например звоним через сипнет на мобильный телефон, который заведомо выключен), в дебаге вижу SIP/2.0 183 Call progress и через какое-то время SIP/2.0 487 Request cancelled. Сипнет говорит, что во время SIP/2.0 183 Call progress проигрывается сообщение, однако * звонящему пиру выдает только длинные гудки... Подскажите, куда ковырять?

Re: Голосовые сообщения об ошибке не приходят с транков

Добавлено: 11 дек 2014, 08:08
virus_net
Убрать r из Dial.

Re: Голосовые сообщения об ошибке не приходят с транков

Добавлено: 11 дек 2014, 08:09
awsswa
снять дамп и посмотреть что на эти сообщения отвечает абонент

Re: Голосовые сообщения об ошибке не приходят с транков

Добавлено: 16 дек 2014, 01:08
gaspaarr
Поубирал везде r, если правильно понимаю, в транк отправляется DIAL_TRUNK_OPTIONS=TL(720000:60000:29500):

Код: Выделить всё

    -- Executing [s@macro-user-callerid:10] Set("SIP/1902-000008d1", "__DIAL_OPTIONS=tL(720000:60000:29500)") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/1902-000008d1", "DIAL_TRUNK_OPTIONS=tL(720000:60000:29500)") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/1902-000008d1", "DIAL_TRUNK_OPTIONS=TL(720000:60000:29500)") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/1902-000008d1", "custom=SIP/SIPNET-1234567") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1902-000008d1", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)TL(720000:60000:29500))") in new stack
    -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/1902-000008d1", "0?Set(DIAL_TRUNK_OPTIONS=TL(720000:60000:29500)M(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:17] Macro("SIP/1902-000008d1", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1902-000008d1", "") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1902-000008d1", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/1902-000008d1", "1?Set(CONNECTEDLINE(num,i)=89161234567)") in new stack
    -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/1902-000008d1", "1?Set(CONNECTEDLINE(name,i)=CID:78621234567)") in new stack
    -- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/1902-000008d1", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:22] Dial("SIP/1902-000008d1", "SIP/SIPNET-1234567/89161234567,300,TL(720000:60000:29500)") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/SIPNET-1234567/89161234567
    -- SIP/SIPNET-1234567-000008d2 is making progress passing it to SIP/1902-000008d1
    -- SIP/SIPNET-1234567-000008d2 is ringing
Лог ответов от абонента астериску, слышу только длинные гудки, потом отбой. Если звонок напрямую через сипнет (без *) -- то сообщение о недоступности номера отлично слышно, потом идет отбой. Может быть проблема в том, что сообщения об ошибке от транка приходят в кодеке, которого нет в *?

Код: Выделить всё

<--- SIP read from UDP:192.168.4.50:5070 --->
INVITE sip:89161234567@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.50:5070;branch=z9hG4bKc9700117
Max-Forwards: 70
To: <sip:89161234567@192.168.1.10>
From: <sip:1902@192.168.1.10>;tag=1714627387
Call-ID: 539e7cb2-f6cd1c1c4fe0d24f8495080023629ff7@192.168.4.50
CSeq: 1 INVITE
Contact: <sip:1902@192.168.4.50:5070>
Supported: replaces
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY,REFER,UPDATE
Content-Type: application/sdp
User-Agent: Panasonic_KX-TGP500B09/22.85 (080023629ff7)
Content-Length: 321

v=0
o=- 1418680182 1418680182 IN IP4 192.168.4.50
s=-
c=IN IP4 192.168.4.50
t=0 0
m=audio 16064 RTP/AVP 9 8 2 18 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
<------------->
--- (13 headers 15 lines) ---
Sending to 192.168.4.50:5070 (NAT)
Sending to 192.168.4.50:5070 (NAT)
Using INVITE request as basis request - 539e7cb2-f6cd1c1c4fe0d24f8495080023629ff7@192.168.4.50
Found peer '1902' for '1902' from 192.168.4.50:5070

<--- Reliably Transmitting (NAT) to 192.168.4.50:5070 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.4.50:5070;branch=z9hG4bKc9700117;received=192.168.4.50;rport=5070
From: <sip:1902@192.168.1.10>;tag=1714627387
To: <sip:89161234567@192.168.1.10>;tag=as7652dbab
Call-ID: 539e7cb2-f6cd1c1c4fe0d24f8495080023629ff7@192.168.4.50
CSeq: 1 INVITE
Server: FPBX-12.0.13(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1830572e"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '539e7cb2-f6cd1c1c4fe0d24f8495080023629ff7@192.168.4.50' in 9792 ms (Method: INVITE)

<--- SIP read from UDP:192.168.4.50:5070 --->
ACK sip:89161234567@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.50:5070;branch=z9hG4bKc9700117
Max-Forwards: 70
To: <sip:89161234567@192.168.1.10>;tag=as7652dbab
From: <sip:1902@192.168.1.10>;tag=1714627387
Call-ID: 539e7cb2-f6cd1c1c4fe0d24f8495080023629ff7@192.168.4.50
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.4.50:5070 --->
INVITE sip:89161234567@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.50:5070;branch=z9hG4bKd6a025c6
Max-Forwards: 70
To: <sip:89161234567@192.168.1.10>
From: <sip:1902@192.168.1.10>;tag=1714627387
Call-ID: 539e7cb2-f6cd1c1c4fe0d24f8495080023629ff7@192.168.4.50
CSeq: 2 INVITE
Contact: <sip:1902@192.168.4.50:5070>
Supported: replaces
Authorization: Digest realm="asterisk", nonce="1830572e", algorithm=MD5, uri="sip:89161234567@192.168.1.10:5060", username="1902", response="863fc1c50c72248abba71541decc3371"
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY,REFER,UPDATE
Content-Type: application/sdp
User-Agent: Panasonic_KX-TGP500B09/22.85 (080023629ff7)
Content-Length: 321

v=0
o=- 1418680182 1418680182 IN IP4 192.168.4.50
s=-
c=IN IP4 192.168.4.50
t=0 0
m=audio 16064 RTP/AVP 9 8 2 18 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
<------------->
--- (14 headers 15 lines) ---
Sending to 192.168.4.50:5070 (NAT)
Using INVITE request as basis request - 539e7cb2-f6cd1c1c4fe0d24f8495080023629ff7@192.168.4.50
Found peer '1902' for '1902' from 192.168.4.50:5070
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format G726-32 for ID 2
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|g726|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g726|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.4.50:16064
Looking for 89161234567 in 4015188-ALL (domain 192.168.1.10)
sip_route_dump: route/path hop: <sip:1902@192.168.4.50:5070>

<--- Transmitting (NAT) to 192.168.4.50:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.50:5070;branch=z9hG4bKd6a025c6;received=192.168.4.50;rport=5070
From: <sip:1902@192.168.1.10>;tag=1714627387
To: <sip:89161234567@192.168.1.10>
Call-ID: 539e7cb2-f6cd1c1c4fe0d24f8495080023629ff7@192.168.4.50
CSeq: 2 INVITE
Server: FPBX-12.0.13(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:89161234567@192.168.1.10:5060>
Content-Length: 0


<------------>


<--- Transmitting (NAT) to 192.168.4.50:5070 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.4.50:5070;branch=z9hG4bKd6a025c6;received=192.168.4.50;rport=5070
From: <sip:1902@192.168.1.10>;tag=1714627387
To: <sip:89161234567@192.168.1.10>;tag=as311e4ca8
Call-ID: 539e7cb2-f6cd1c1c4fe0d24f8495080023629ff7@192.168.4.50
CSeq: 2 INVITE
Server: FPBX-12.0.13(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:89161234567@192.168.1.10:5060>
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 192.168.4.50:5070 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.4.50:5070;branch=z9hG4bKd6a025c6;received=192.168.4.50;rport=5070
From: <sip:1902@192.168.1.10>;tag=1714627387
To: <sip:89161234567@192.168.1.10>;tag=as311e4ca8
Call-ID: 539e7cb2-f6cd1c1c4fe0d24f8495080023629ff7@192.168.4.50
CSeq: 2 INVITE
Server: FPBX-12.0.13(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:89161234567@192.168.1.10:5060>
Content-Length: 0


<------------>
Reliably Transmitting (NAT) to 192.168.4.50:5070:
OPTIONS sip:1902@192.168.4.50:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK6607f2ea;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.10>;tag=as25f3a89d
To: <sip:1902@192.168.4.50:5070>
Contact: <sip:Unknown@192.168.1.10:5060>
Call-ID: 16d23ea136f5dc185ac896f82195eb39@192.168.1.10:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.13(13.0.1)
Date: Mon, 15 Dec 2014 21:49:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.4.50:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK6607f2ea;rport=5060
To: <sip:1902@192.168.4.50>;tag=2217326354
From: "Unknown" <sip:Unknown@192.168.1.10>;tag=as25f3a89d
Call-ID: 16d23ea136f5dc185ac896f82195eb39@192.168.1.10:5060
CSeq: 102 OPTIONS
Contact: <sip:192.168.4.50:5070>
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY,REFER,UPDATE
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '16d23ea136f5dc185ac896f82195eb39@192.168.1.10:5060' Method: OPTIONS

<--- Reliably Transmitting (NAT) to 192.168.4.50:5070 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.4.50:5070;branch=z9hG4bKd6a025c6;received=192.168.4.50;rport=5070
From: <sip:1902@192.168.1.10>;tag=1714627387
To: <sip:89161234567@192.168.1.10>;tag=as311e4ca8
Call-ID: 539e7cb2-f6cd1c1c4fe0d24f8495080023629ff7@192.168.4.50
CSeq: 2 INVITE
Server: FPBX-12.0.13(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<------------>
[2014-12-16 00:49:50] WARNING[14303][C-000019c8]: channel.c:4816 ast_prod: Prodding channel 'SIP/1902-000008cf' failed

<--- SIP read from UDP:192.168.4.50:5070 --->
ACK sip:89161234567@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.50:5070;branch=z9hG4bKd6a025c6
Max-Forwards: 70
To: <sip:89161234567@192.168.1.10>;tag=as311e4ca8
From: <sip:1902@192.168.1.10>;tag=1714627387
Call-ID: 539e7cb2-f6cd1c1c4fe0d24f8495080023629ff7@192.168.4.50
CSeq: 2 ACK
Authorization: Digest realm="asterisk", nonce="1830572e", algorithm=MD5, uri="sip:89161234567@192.168.1.10:5060", username="1902", response="d85286d81df22aaf57c6d9eba6c22e00"
Content-Length: 0

<------------>

Re: Голосовые сообщения об ошибке не приходят с транков

Добавлено: 16 дек 2014, 08:20
virus_net
В транк отправляется:
gaspaarr писал(а):SIP/SIPNET-1234567/89161234567,300,TL(720000:60000:29500)
исходя и этого вывода можно судить что "ДА", r вы убрали.
gaspaarr писал(а):Может быть проблема в том, что....
Что бы это понять нужно дебажить оба "плеча". Вы пока посмотрели только одно - астериск-ваш абонент и мне пока не понятна ваша логика при ответе на вопрос "с чего начать?".
Ведь то что вам надо должно приходить со стороны VoIP оператора.
gaspaarr писал(а):Лог ответов от абонента астериску
А нужно "лог" от оператора к астериску, т.к. именно оттуда должно приходить early media.

+ уже было сказано:
снимите дамп и анализируйте его

Re: Голосовые сообщения об ошибке не приходят с транков

Добавлено: 16 дек 2014, 20:26
gaspaarr
Дамп от оператора смотрел... Насколько я понял, * не транслирует клиенту early media, а чтобы клиент его получал, надо разрешать directmedia=yes но тогда не будет записи разговоров... ошибаюсь?

Код: Выделить всё

Audio is at 19118
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 212.53.40.40:5060:
INVITE sip:89161234567@sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 19.5.9.11:5060;branch=z9hG4bK3b42631c;rport
Max-Forwards: 70
From: <sip:12345678901@sipnet.ru>;tag=as5d063a39
To: <sip:89161234567@sipnet.ru>
Contact: <sip:12345678901@19.5.9.11:5060>
Call-ID: 46974afc03eba7e12b4f660842b6d4e2@sipnet.ru
CSeq: 102 INVITE
User-Agent: FPBX-12.0.13(13.0.1)
Date: Tue, 16 Dec 2014 17:21:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 267126869 267126869 IN IP4 19.5.9.11
s=Asterisk PBX 13.0.1
c=IN IP4 19.5.9.11
t=0 0
m=audio 19118 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:212.53.40.40:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 19.5.9.11:5060;branch=z9hG4bK3b42631c;rport=5060
From: <sip:12345678901@sipnet.ru>;tag=as5d063a39
To: <sip:89161234567@sipnet.ru>
Call-ID: 46974afc03eba7e12b4f660842b6d4e2@sipnet.ru
CSeq: 102 INVITE
Server: CommuniGatePro/6.0.11
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:212.53.40.40:5060 --->
SIP/2.0 401 Authentication required
Via: SIP/2.0/UDP 19.5.9.11:5060;branch=z9hG4bK3b42631c;rport=5060
From: <sip:12345678901@sipnet.ru>;tag=as5d063a39
To: <sip:89161234567@sipnet.ru>;tag=836C484C
Call-ID: 46974afc03eba7e12b4f660842b6d4e2@sipnet.ru
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="etc.tario.ru",nonce="",opaque="opaq",qop="auth",algorithm=MD5
Server: CommuniGatePro/6.0.11
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 212.53.40.40:5060:
ACK sip:89161234567@sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 19.5.9.11:5060;branch=z9hG4bK3b42631c;rport
Max-Forwards: 70
From: <sip:12345678901@sipnet.ru>;tag=as5d063a39
To: <sip:89161234567@sipnet.ru>;tag=836C484C
Contact: <sip:12345678901@19.5.9.11:5060>
Call-ID: 46974afc03eba7e12b4f660842b6d4e2@sipnet.ru
CSeq: 102 ACK
User-Agent: FPBX-12.0.13(13.0.1)
Content-Length: 0


---
Audio is at 19118
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 212.53.40.40:5060:
INVITE sip:89161234567@sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 19.5.9.11:5060;branch=z9hG4bK667ff51d;rport
Max-Forwards: 70
From: <sip:12345678901@sipnet.ru>;tag=as5d063a39
To: <sip:89161234567@sipnet.ru>
Contact: <sip:12345678901@19.5.9.11:5060>
Call-ID: 46974afc03eba7e12b4f660842b6d4e2@sipnet.ru
CSeq: 103 INVITE
User-Agent: FPBX-12.0.13(13.0.1)
Authorization: Digest username="1234567", realm="etc.tario.ru", algorithm=MD5, uri="sip:89161234567@sipnet.ru", nonce="", response="", opaque="opaq", qop=auth, cnonce="27fcf3bf", nc=00000001
Date: Tue, 16 Dec 2014 17:21:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 267126869 267126870 IN IP4 19.5.9.11
s=Asterisk PBX 13.0.1
c=IN IP4 19.5.9.11
t=0 0
m=audio 19118 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:212.53.40.40:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 19.5.9.11:5060;branch=z9hG4bK667ff51d;rport=5060
From: <sip:12345678901@sipnet.ru>;tag=as5d063a39
To: <sip:89161234567@sipnet.ru>
Call-ID: 46974afc03eba7e12b4f660842b6d4e2@sipnet.ru
CSeq: 103 INVITE
Server: CommuniGatePro/6.0.11
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:212.53.40.40:5060 --->
SIP/2.0 183 Call progress
Via: SIP/2.0/UDP 19.5.9.11:5060;branch=z9hG4bK667ff51d;rport=5060
Record-Route: <sip:1887755-192.168.40.71.dialog.cgatepro;lr>
Record-Route: <sip:192.168.40.71:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:12345678901@sipnet.ru>;tag=as5d063a39
To: <sip:89161234567@sipnet.ru>;tag=09DEB1FE-64528-0C821828_kmbdctj-EF93
Call-ID: 46974afc03eba7e12b4f660842b6d4e2@sipnet.ru
CSeq: 103 INVITE
Contact: <sip:signode-64528-0C821828_kmbdctj-EF93@212.53.40.40>
Supported: 100rel,timer,replaces,histinfo,precondition
User-Agent: CommuniGatePro-callLeg/6.0.11
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,REFER
Content-Type: application/sdp
Content-Length: 334

v=0
o=CGPLeg064528 4207824180 2103912091 IN IP4 212.53.40.71
s=-
c=IN IP4 212.53.40.71
t=0 0
m=audio 23784 RTP/AVP 0 101
c=IN IP4 212.53.40.71
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=ice-pwd:C28CAB2990E57AF2AE9DA5EE
a=ice-ufrag:pf0642910
a=silenceSupp:off - - - -
<------------->
--- (15 headers 15 lines) ---
sip_route_dump: route/path hop: <sip:212.53.40.40:5060;lr>
sip_route_dump: route/path hop: <sip:192.168.40.71:5060;lr>
sip_route_dump: route/path hop: <sip:1887755-192.168.40.71.dialog.cgatepro;lr>
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw|g729), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 212.53.40.71:23784

<--- SIP read from UDP:212.53.40.40:5060 --->
SIP/2.0 183 Call progress
Via: SIP/2.0/UDP 19.5.9.11:5060;branch=z9hG4bK667ff51d;rport=5060
Record-Route: <sip:1887755-192.168.40.71.dialog.cgatepro;lr>
Record-Route: <sip:192.168.40.71:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:12345678901@sipnet.ru>;tag=as5d063a39
To: <sip:89161234567@sipnet.ru>;tag=09DEB1FE-64528-0C821828_kmbdctj-EF93
Call-ID: 46974afc03eba7e12b4f660842b6d4e2@sipnet.ru
CSeq: 103 INVITE
Contact: <sip:signode-64528-0C821828_kmbdctj-EF93@212.53.40.40>
Supported: 100rel,timer,replaces,histinfo,precondition
User-Agent: CommuniGatePro-callLeg/6.0.11
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,REFER
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:212.53.40.40:5060;lr>
sip_route_dump: route/path hop: <sip:192.168.40.71:5060;lr>
sip_route_dump: route/path hop: <sip:1887755-192.168.40.71.dialog.cgatepro;lr>
sip*CLI>

<--- SIP read from UDP:212.53.40.40:5060 --->
SIP/2.0 487 Request cancelled
Via: SIP/2.0/UDP 19.5.9.11:5060;branch=z9hG4bK667ff51d;rport=5060
From: <sip:12345678901@sipnet.ru>;tag=as5d063a39
To: <sip:89161234567@sipnet.ru>;tag=09DEB1FE-64528-0C821828_kmbdctj-EF93
Call-ID: 46974afc03eba7e12b4f660842b6d4e2@sipnet.ru
CSeq: 103 INVITE
Server: CommuniGatePro/6.0.11
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 212.53.40.40:5060:
ACK sip:signode-64528-0C821828_kmbdctj-EF93@212.53.40.40 SIP/2.0
Via: SIP/2.0/UDP 19.5.9.11:5060;branch=z9hG4bK667ff51d;rport
Max-Forwards: 70
From: <sip:12345678901@sipnet.ru>;tag=as5d063a39
To: <sip:89161234567@sipnet.ru>;tag=09DEB1FE-64528-0C821828_kmbdctj-EF93
Contact: <sip:12345678901@19.5.9.11:5060>
Call-ID: 46974afc03eba7e12b4f660842b6d4e2@sipnet.ru
CSeq: 103 ACK
User-Agent: FPBX-12.0.13(13.0.1)
Content-Length: 0

Re: Голосовые сообщения об ошибке не приходят с транков

Добавлено: 16 дек 2014, 20:58
ded
https://wiki.asterisk.org/wiki/display/ ... pplication

Re: Голосовые сообщения об ошибке не приходят с транков

Добавлено: 16 дек 2014, 23:15
gaspaarr
Проблема решилась указанием progressinband=no в sip conf general, не совсем понятно, почему по умолчанию указано never:

Код: Выделить всё

progressinband=yes

When "RING" event is requested, always send 180 Ringing (if it hasn't been sent yet) followed by 183 Session Progress and in-band audio

progressinband=no

Send 180 Ringing if 183 has not yet been sent establishing audio path. If audio path is established already (with 183) then send in-band ringing (this is the way asterisk historically behaved because of buggy phones like polycom)

progressinband=never

Whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. [b]This is the default behaviour of Asterisk.[/b]

Re: Голосовые сообщения об ошибке не приходят с транков

Добавлено: 17 дек 2014, 11:43
gaspaarr
Правда теперь появилась другая проблема, разговоры начинают записываться не с момента ответа, а сразу после набора номера и начала гудков...

Re: Голосовые сообщения об ошибке не приходят с транков

Добавлено: 17 дек 2014, 18:17
virus_net
для решения этой "проблемы" у Dial есть магические опции:
M(x): Executes the macro (x) upon connect of the call (i.e. when the called party answers).
U(x): Executes, via gosub, routine x on the called channel. This is similar to M above, but a gosub rather than a macro.
Часовой пояс: UTC+03:00
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