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Проблема звонка

Обрыв звонка
<1234 5 678>
allel
Avatara of allel
Откуда: Курск
Сообщений: 29

Re: Проблема звонка

Это вывод CLI как только беру трубку связь рвется
-- Executing [89202615330@numberplan-custom-2:1] Macro("SIP/55330-b580b0a8", "trunkdial|SIP/trunk_3/89202615330|") in new stack
-- Executing [s@macro-trunkdial:1] Set("SIP/55330-b580b0a8", "CALLERID(all)=") in new stack
-- Executing [s@macro-trunkdial:2] Dial("SIP/55330-b580b0a8", "SIP/trunk_3/89202615330") in new stack
-- Called trunk_3/89202615330
-- SIP/trunk_3-08269f88 is making progress passing it to SIP/55330-b580b0a8
-- SIP/trunk_3-08269f88 is making progress passing it to SIP/55330-b580b0a8
-- SIP/trunk_3-08269f88 is ringing
-- SIP/trunk_3-08269f88 is making progress passing it to SIP/55330-b580b0a8
-- SIP/trunk_3-08269f88 answered SIP/55330-b580b0a8
-- Native bridging SIP/55330-b580b0a8 and SIP/trunk_3-08269f88
== Spawn extension (macro-trunkdial, s, 2) exited non-zero on 'SIP/55330-b580b0a8' in macro 'trunkdial'
== Spawn extension (macro-trunkdial, s, 2) exited non-zero on 'SIP/55330-b580b0a8'
Настройки провайдера щас малех позднее
2008-09-10 16:15

betman
Откуда: Kiev
Сообщений: 801

Re: Проблема звонка

Ivon:

а с русским-то языком проблемы у Вас, так как читаете слова на свой лад, как Вам в настоящий момент хочется.
Ну написано: betman,kiev (Украина, планета Земля, Млечный путь). Русский застал в 1 классе -- дальше перестройка и т.п. Могу и на украинском -- грамотно (на 5, как в табеле). Но думаю, не поймете....

Лучший способ предвидеть будущее - изобрести его (Алан Кей, "Apple")
2008-09-10 16:22

betman
Откуда: Kiev
Сообщений: 801

Re: Проблема звонка

Ivon:

Ну, кто-то вдумывается в смысл слов, а кто-то имеет пачку шаблонов, и умеет копипастить, а моск держит выключеным до лучших времен.
to betman: а с русским-то языком проблемы у Вас, так как читаете слова на свой лад, как Вам в настоящий момент хочется. "мне уже 18 и я постоянно читаю www.udaff.com, тем и выделяюсь" - эту стадию Вы видимо уже завершаете. Дык вот, я изначально пошел другим путем, так что оно ("мне уже 18 и я постоянно читаю www.udaff.com, тем и выделяюсь") меня не коснулось.
.... а в целом -- спор в песочнице.....
Лучший способ предвидеть будущее - изобрести его (Алан Кей, "Apple")
2008-09-10 16:23

Ivon
Avatara of Ivon
Сообщений: 445

Re: Проблема звонка

В CLI астериска вводим команду: sip debug peer trunk_3
убеждаемся, что команда принята,
набираем номер,
в консоли появляется много информации,
постим эту информацию в эту тему,
а иначе, будете еще долго шокироваться от "многозначительных" выражений ала Винни Пух: "Это сообщение само по себе не появляется ...."
2008-09-10 16:25

Ivon
Avatara of Ivon
Сообщений: 445

Re: Проблема звонка

to betman: Я тоже не собириаюсь больше спорить, просто хотел подсказать кратчайший путь, национальной неприязни не имею, все, заканчиваю флейм в этой теме
2008-09-10 16:28

allel
Avatara of allel
Откуда: Курск
Сообщений: 29

Re: Проблема звонка

Вот

SIP Debugging Enabled for IP: 212.53.40.40:5060
The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.
-- Executing [89202615330@numberplan-custom-2:1] Macro("SIP/55330-b580c638", "trunkdial|SIP/trunk_3/89202615330|") in new stack
-- Executing [s@macro-trunkdial:1] Set("SIP/55330-b580c638", "CALLERID(all)=") in new stack
-- Executing [s@macro-trunkdial:2] Dial("SIP/55330-b580c638", "SIP/trunk_3/89202615330") in new stack
Audio is at ---.---.---.--- port 17152
Adding codec 0x800 (g726) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
INVITE sip:89202615330@212.53.40.40 SIP/2.0
Via: SIP/2.0/UDP ---.---.---.---:5060;branch=z9hG4bK3cbcc8e4;rport
From: "asterisk" <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>
Contact: <sip:4350754@---.---.---.--->
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Sep 2008 12:26:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 496
asterisk*CLI>
v=0
o=root 2978 2978 IN IP4 ---.---.---.---
s=session
c=IN IP4 ---.---.---.---
t=0 0
m=audio 17152 RTP/AVP 2 3 0 8 112 5 10 7 110 97 101
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called trunk_3/89202615330
asterisk*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 92.50.206.238:5060;branch=z9hG4bK3cbcc8e4;rport=5060
From: "asterisk" <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 102 INVITE
Server: CommuniGatePro/5.2.8a
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
asterisk*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 401 Authentication required
Via: SIP/2.0/UDP ---.---.---.---:5060;branch=z9hG4bK3cbcc8e4;rport=5060
From: "asterisk" <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=5660BEE0
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="etc.tario.ru",nonce="05AF0A3525B8E93C8A7E",opaque="opaqueData",qop="auth",algorithm=MD5
Server: CommuniGatePro/5.2.8a
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 212.53.40.40:5060:
ACK sip:89202615330@212.53.40.40 SIP/2.0
Via: SIP/2.0/UDP ---.---.---.---:5060;branch=z9hG4bK3cbcc8e4;rport
From: "asterisk" <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=5660BEE0
Contact: <sip:4350754@---.---.---.--->
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Audio is at ---.---.---.--- port 17152
Adding codec 0x800 (g726) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
INVITE sip:89202615330@212.53.40.40 SIP/2.0
Via: SIP/2.0/UDP ---.---.---.---:5060;branch=z9hG4bK535ba269;rport
From: "asterisk" <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>
Contact: <sip:4350754@---.---.---.--->
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="4350754", realm="etc.tario.ru", algorithm=MD5, uri="sip:89202615330@212.53.40.40", nonce="05AF0A3525B8E93C8A7E", response="44c3098f16cc2d63178fff302dc8b7d8", opaque="opaqueData", qop=auth, cnonce="45edd293", nc=00000001
Date: Wed, 10 Sep 2008 12:26:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 496

v=0
o=root 2978 2979 IN IP4 92.50.206.238
s=session
c=IN IP4 ---.---.---.---
t=0 0
m=audio 17152 RTP/AVP 2 3 0 8 112 5 10 7 110 97 101
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
asterisk*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ---.---.---.---:5060;branch=z9hG4bK535ba269;rport=5060
From: "asterisk" <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 103 INVITE
Server: CommuniGatePro/5.2.8a
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
asterisk*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ---.---.---.---:5060;rport=5060;branch=z9hG4bK535ba269
Record-Route: <sip:212.53.35.244:5060;lr>,<sip:285379-212.53.40.73.dialog.cgatepro;lr>
Record-Route: <sip:212.53.40.73:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=38be6af0-3025566
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 103 INVITE
Content-Type: application/sdp
Server: TarioSoftswitch/3.2.10
Content-Length: 179

v=0
o=Tario-Softswitch 30259 101 IN IP4 212.53.40.94
s=SIP Call
c=IN IP4 212.53.40.94
t=0 0
m=audio 27092 RTP/AVP 8
c=IN IP4 212.53.40.94
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 212.53.40.94:27092
Found audio description format PCMA for ID 8
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 212.53.40.94:27092
-- SIP/trunk_3-08269f88 is making progress passing it to SIP/55330-b580c638
asterisk*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ---.---.---.---:5060;rport=5060;branch=z9hG4bK535ba269
Record-Route: <sip:212.53.35.244:5060;lr>,<sip:285379-212.53.40.73.dialog.cgatepro;lr>
Record-Route: <sip:212.53.40.73:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=38be6af0-3025566
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 103 INVITE
Content-Type: application/sdp
Server: TarioSoftswitch/3.2.10
Content-Length: 179

v=0
o=Tario-Softswitch 30259 101 IN IP4 212.53.40.94
s=SIP Call
c=IN IP4 212.53.40.94
t=0 0
m=audio 27092 RTP/AVP 8
c=IN IP4 212.53.40.94
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 212.53.40.94:27092
Found audio description format PCMA for ID 8
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 212.53.40.94:27092
-- SIP/trunk_3-08269f88 is making progress passing it to SIP/55330-b580c638
asterisk*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ---.---.---.---:5060;rport=5060;branch=z9hG4bK535ba269
Record-Route: <sip:212.53.35.244:5060;lr>,<sip:285379-212.53.40.73.dialog.cgatepro;lr>
Record-Route: <sip:212.53.40.73:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=38be6af0-3025566
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 103 INVITE
Content-Type: application/sdp
Server: TarioSoftswitch/3.2.10
Content-Length: 179

v=0
o=Tario-Softswitch 30259 101 IN IP4 212.53.40.94
s=SIP Call
c=IN IP4 212.53.40.94
t=0 0
m=audio 27092 RTP/AVP 8
c=IN IP4 212.53.40.94
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 212.53.40.94:27092
Found audio description format PCMA for ID 8
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 212.53.40.94:27092
-- SIP/trunk_3-08269f88 is ringing
-- SIP/trunk_3-08269f88 is making progress passing it to SIP/55330-b580c638
asterisk*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ---.---.---.---:5060;rport=5060;branch=z9hG4bK535ba269
Record-Route: <sip:212.53.35.244:5060;lr>,<sip:285379-212.53.40.73.dialog.cgatepro;lr>
Record-Route: <sip:212.53.40.73:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=38be6af0-3025566
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 103 INVITE
Content-Type: application/sdp
Server: TarioSoftswitch/3.2.10
Content-Length: 179

v=0
o=Tario-Softswitch 30259 102 IN IP4 212.53.40.81
s=SIP Call
c=IN IP4 212.53.40.81
t=0 0
m=audio 23860 RTP/AVP 8
c=IN IP4 212.53.40.81
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 212.53.40.81:23860
Found audio description format PCMA for ID 8
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 212.53.40.81:23860
-- SIP/trunk_3-08269f88 is making progress passing it to SIP/55330-b580c638
asterisk*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ---.---.---.---:5060;rport=5060;branch=z9hG4bK535ba269
Record-Route: <sip:212.53.35.244:5060;lr>,<sip:285379-212.53.40.73.dialog.cgatepro;lr>
Record-Route: <sip:212.53.40.73:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=38be6af0-3025566
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 103 INVITE
Content-Type: application/sdp
Server: TarioSoftswitch/3.2.10
Content-Length: 179

v=0
o=Tario-Softswitch 30259 102 IN IP4 212.53.40.81
s=SIP Call
c=IN IP4 212.53.40.81
t=0 0
m=audio 23860 RTP/AVP 8
c=IN IP4 212.53.40.81
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 212.53.40.81:23860
Found audio description format PCMA for ID 8
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 212.53.40.81:23860
-- SIP/trunk_3-08269f88 is making progress passing it to SIP/55330-b580c638
asterisk*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ---.---.---.---:5060;rport=5060;branch=z9hG4bK535ba269
Record-Route: <sip:212.53.35.244:5060;lr>,<sip:285379-212.53.40.73.dialog.cgatepro;lr>
Record-Route: <sip:212.53.40.73:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=38be6af0-3025566
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 103 INVITE
Content-Type: application/sdp
Server: TarioSoftswitch/3.2.10
Content-Length: 179

v=0
o=Tario-Softswitch 30259 102 IN IP4 212.53.40.81
s=SIP Call
c=IN IP4 212.53.40.81
t=0 0
m=audio 23860 RTP/AVP 8
c=IN IP4 212.53.40.81
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 212.53.40.81:23860
Found audio description format PCMA for ID 8
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 212.53.40.81:23860
-- SIP/trunk_3-08269f88 is ringing
-- SIP/trunk_3-08269f88 is making progress passing it to SIP/55330-b580c638
asterisk*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ---.---.---.---:5060;rport=5060;branch=z9hG4bK535ba269
Record-Route: <sip:212.53.35.244:5060;lr>,<sip:285379-212.53.40.73.dialog.cgatepro;lr>
Record-Route: <sip:212.53.40.73:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=38be6af0-3025566
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 103 INVITE
Contact: <sip:proc-3025566@212.53.35.244>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, INFO, OPTIONS
Server: TarioSoftswitch/3.2.10
Content-Length: 179

v=0
o=Tario-Softswitch 30259 102 IN IP4 212.53.40.81
s=SIP Call
c=IN IP4 212.53.40.81
t=0 0
m=audio 23860 RTP/AVP 8
c=IN IP4 212.53.40.81
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (14 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 212.53.40.81:23860
Found audio description format PCMA for ID 8
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 212.53.40.81:23860
Transmitting (no NAT) to 212.53.40.40:5060:
ACK sip:proc-3025566@212.53.35.244 SIP/2.0
Via: SIP/2.0/UDP ---.---.---.---:5060;branch=z9hG4bK578f9655;rport
Route: <sip:212.53.40.40:5060;lr>,<sip:212.53.40.73:5060;lr>,<sip:285379-212.53.40.73.dialog.cgatepro;lr>,<sip:212.53.35.244:5060;lr>
From: "asterisk" <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=38be6af0-3025566
Contact: <sip:4350754@---.---.---.--->
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/trunk_3-08269f88 answered SIP/55330-b580c638
-- Native bridging SIP/55330-b580c638 and SIP/trunk_3-08269f88
set_destination: Parsing <sip:212.53.40.40:5060;lr> for address/port to send to
set_destination: set destination to 212.53.40.40, port 5060
Audio is at ---.---.---.--- port 17152
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
INVITE sip:proc-3025566@212.53.35.244 SIP/2.0
Via: SIP/2.0/UDP ---.---.---.---:5060;branch=z9hG4bK3dd57100;rport
Route: <sip:212.53.40.40:5060;lr>,<sip:212.53.40.73:5060;lr>,<sip:285379-212.53.40.73.dialog.cgatepro;lr>,<sip:212.53.35.244:5060;lr>
From: "asterisk" <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=38be6af0-3025566
Contact: <sip:4350754@---.---.---.--->
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 182

v=0
o=root 2978 2980 IN IP4 ---.---.---.---
s=session
c=IN IP4 ---.---.---.---
t=0 0
m=audio 16436 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
asterisk*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ---.---.---.---:5060;branch=z9hG4bK3dd57100;rport=5060
From: "asterisk" <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=38be6af0-3025566
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 104 INVITE
Server: CommuniGatePro/5.2.8a
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
asterisk*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP ---.---.---.---:5060;rport=5060;branch=z9hG4bK3dd57100
From: <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=38be6af0-3025566
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 104 INVITE
Server: TarioSoftswitch/3.2.10
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:212.53.40.40:5060;lr> for address/port to send to
set_destination: set destination to 212.53.40.40, port 5060
Transmitting (no NAT) to 212.53.40.40:5060:
ACK sip:proc-3025566@212.53.35.244 SIP/2.0
Via: SIP/2.0/UDP ---.---.---.---:5060;branch=z9hG4bK3dd57100;rport
Route: <sip:212.53.40.40:5060;lr>,<sip:212.53.40.73:5060;lr>,<sip:285379-212.53.40.73.dialog.cgatepro;lr>,<sip:212.53.35.244:5060;lr>
From: "asterisk" <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=38be6af0-3025566
Contact: <sip:4350754@---.---.---.--->
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog '1a7ecfe970c2078a691d571a65e831ec@212.53.40.40' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:212.53.40.40:5060;lr> for address/port to send to
set_destination: set destination to 212.53.40.40, port 5060
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
BYE sip:proc-3025566@212.53.35.244 SIP/2.0
Via: SIP/2.0/UDP ---.---.---.---:5060;branch=z9hG4bK43bef360;rport
Route: <sip:212.53.40.40:5060;lr>,<sip:212.53.40.73:5060;lr>,<sip:285379-212.53.40.73.dialog.cgatepro;lr>,<sip:212.53.35.244:5060;lr>
From: "asterisk" <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=38be6af0-3025566
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="4350754", realm="etc.tario.ru", algorithm=MD5, uri="sip:proc-3025566@212.53.35.244", nonce="05AF0A3525B8E93C8A7E", response="d14b6623ffdae1945f11aebec656d21b", opaque="opaqueData", qop=auth, cnonce="77fe176d", nc=00000002
Content-Length: 0


---
== Spawn extension (macro-trunkdial, s, 2) exited non-zero on 'SIP/55330-b580c638' in macro 'trunkdial'
== Spawn extension (macro-trunkdial, s, 2) exited non-zero on 'SIP/55330-b580c638'
asterisk*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ---.---.---.---:5060;rport=5060;branch=z9hG4bK43bef360
From: <sip:4350754@212.53.40.40>;tag=as52879bc3
To: <sip:89202615330@212.53.40.40>;tag=38be6af0-3025566
Call-ID: 1a7ecfe970c2078a691d571a65e831ec@212.53.40.40
CSeq: 105 BYE
Allow: INVITE, ACK, BYE, CANCEL, INFO, OPTIONS
Server: TarioSoftswitch/3.2.10
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '1a7ecfe970c2078a691d571a65e831ec@212.53.40.40' Method: INVITE
2008-09-10 16:37

cron333
Сообщений: 1573

Re: Проблема звонка

Как у вас организован выход в интернет? Есть - NAT? Настроен файервол?
Вызов идет через сипнет. Этот пров требует обязательной регистрации. Без нее не будет "звонить"
2008-09-10 16:54

cron333
Сообщений: 1573

Re: Проблема звонка

Ivon:

а иначе, будете еще долго шокироваться от "многозначительных" выражений ала Винни Пух: "Это сообщение само по себе не появляется ...."
Уважаемый, вас похоже понесло ...
Для вас лично: ни одно сообщение само по себе не появляется ... (может, если только при ваших настройках)

Вы же советуете держать "моск" включенным. Почему сами этим не пользуетесь?
2008-09-10 16:59

allel
Avatara of allel
Откуда: Курск
Сообщений: 29

Re: Проблема звонка

Выход в инет прямой ната нету фаервол пропускает всё с домена сипнета. Регистрация есть на вопрос sip show registry отвечает registred
2008-09-10 17:01

allel
Avatara of allel
Откуда: Курск
Сообщений: 29

Re: Проблема звонка

Проблема реально появилась после втыкания карточки пробовал убрать карту с зачисткой ее сведений и звонки начинают ходить нормально
2008-09-10 17:03

<1234 5 678>
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