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проблема с call-файлами и originate

0

Asterisk подключен к SIP серверу муль-тифона вот sip.conf

[general]
tcpenable=yes

allow=all
sendrpid=yes 
register=>7929*******@multifon.ru:password:7929*******@193.201.229.35:5060/7929*******

[multifon-out]
dtmfmode=inband
username=7929*******
type=peer
secret=password
host=193.201.229.35
fromuser=7929*******
fromdomain=multifon.ru
port=5060
nat=yes
context=incoming
insecure=port,invite
insecure=invite
sendrpid=yes

extensions.conf

[incoming]
exten => 7929*******,1,Answer
exten => 7929*******,n,Wait(10)
exten => 7929*******,n,Hangup


[out_context]
exten => s,1,AGI(agi_test.php,${text})

[out]
exten=>_7XXXXXXXXXX,1,Dial(SIP/${EXTEN}@multifon-out,30,r)

[local]
exten=>_1XX,1,Dial(SIP/${EXTEN},60,rt) 

[phones]
include => incoming
include => local
include => out

Вот собственно call-файл через который пытаюсь инициировать вызов через SIP/multifon-out:

Channel: SIP/79299999999@multifon-out
MaxRetries: 1
RetryTime: 30
WaitTime: 30
Context: out_context
Extension: s
Set: text=proverka
Priority: 1

В cli вываливается следующее:

  -- Attempting call on SIP/79299999999@multifon-out for s@out_context:1 (Retry 1)
  == Using SIP RTP CoS mark 5
[Feb  7 15:54:54] NOTICE[7593]: pbx_spool.c:369 attempt_thread: Call failed to go through, reason (1) Hangup
    -- Attempting call on SIP/79299999999@multifon-out for s@out_context:1 (Retry 2)
  == Using SIP RTP CoS mark 5
[Feb  7 15:55:24] NOTICE[7594]: pbx_spool.c:369 attempt_thread: Call failed to go through, reason (1) Hangup
[Feb  7 15:55:24] NOTICE[7594]: pbx_spool.c:372 attempt_thread: Queued call to SIP/79299999999@multifon-out expired without completion after 1 attempt

Если инициировать через AMI методом Originate() то выдаёт следующее:

  == Manager 'callnotice' logged on from 192.168.1.222
  == Using SIP RTP CoS mark 5
  == Manager 'callnotice' logged off from 192.168.1.222
[Feb  7 15:57:16] WARNING[7572][C-0000000a]: chan_sip.c:22376 handle_response_invite: Received response: "Forbidden" from '"Anonymous" <sip:7929*******@anonymous.invalid>;tag=as79a82160'

Хотя нужно сказать, что если добавить в extensions.conf в контекст [incoming] следующуюю строку: exten=> 7929*,1,Dial(SIP/79299999999@multifon-out,30,r) произвести вызов на номер 7929* то Asterisk успешно наберет 79299999999. Подскажите куда копать, чтобы сделать исходящий звонок через call-файл или originate. Спасибо.

удалить закрыть спам изменить тег редактировать

спросил 2013-02-07 13:05:11 +0400

tlq Gravatar tlq
1 1 1

Comments

Обработкой переменной text и последующей выдачей аудиофайла в эфир.

tlq ( 2013-02-07 15:27:18 +0400 )редактировать

4 Ответа

0

А чем у тебя занимается "exten => s,1,AGI(agi_test.php,${text})"?

ссылка удалить спам редактировать

ответил 2013-02-07 15:23:47 +0400

bolshoy_plohish Gravatar bolshoy_plohish
1358 21 16 37

Comments

Вызов внутреннего или внешнего екстеншена и соединение его с AGI скриптом для отправки

сообщения.

Channel: Local/1000@from-internal

MaxRetries: 0

RetryTime: 15

WaitTime: 15

Application: AGI

Data: myagi.agi

bolshoy_plohish ( 2013-02-07 15:59:29 +0400 )редактировать
0

А сами пробовали позвонить через SIP/79299999999@multifon-out??? скорее всего мультифон не принимает ваш звонок из-за отсутствия нужного CID. А вообще тема dialout настолько истоптана что найти информацию по ней не проблема. Копайте в сторону гугла.

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ответил 2013-02-07 13:42:42 +0400

switch Gravatar switch
8334 11 7 91
http://lynks.ru/
0
asterisk -r

sip set debug on
ссылка удалить спам редактировать

ответил 2013-02-07 14:16:48 +0400

meral Gravatar meral flag of Ukraine
21228 24 19 169
http://pro-sip.net/
0

Попробовал установить Callerid: 7929* <7929*******> - ничего не происходит :(. Через SIP/79299999999@multifon-out звонок проходит.

[Feb  7 17:40:48] VERBOSE[7671] pbx_spool.c:     -- Attempting call on SIP/79299999999@multifon-out for s@out_context:1 (Retry 1)
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] netsock2.c:   == Using SIP RTP CoS mark 5
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: We think we can do text
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Audio is at 14672
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100019 (slin) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100001 (g723) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100002 (gsm) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100002 (gsm) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100003 (ulaw) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100003 (ulaw) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100004 (alaw) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100004 (alaw) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100005 (g726aal2) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100006 (adpcm) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100007 (lpc10) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100008 (g729) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100009 (speex) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100010 (ilbc) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100011 (g726) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100012 (g722) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100013 (siren7) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100014 (siren14) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100015 (g719) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100016 (speex16) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100017 (testlaw) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100017 (testlaw) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100020 (slin12) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100021 (slin16) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100022 (slin24) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100023 (slin32) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100024 (slin44) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100025 (slin48) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100026 (slin96) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100027 (slin192) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Adding codec 100028 (speex32) to SDP
    [Feb  7 17:40:48] VERBOSE[7671][C-00000023] chan_sip.c: Reliably Transmitting (NAT) to 193.201.229.35:5060:
    INVITE sip:79299999999@193.201.229.35:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.222:5060;branch=z9hG4bK1cb00090;rport
    Max-Forwards: 70
    From: "7929*******" <sip:7929*******@multifon.ru>;tag=as59f41166
    To: <sip:79299999999@193.201.229.35:5060>
    Contact: <sip:7929*******@192.168.1.222:5060>
    Call-ID: 32c8275544b26a577d38a7ad072a4ef6@multifon.ru
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 11.0.1
    Date: Thu, 07 Feb 2013 10:40:48 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Remote-Party-ID: "7929*******" <sip:7929*******@multifon.ru>;party=calling;privacy=off;screen=no
    Content-Type: application/sdp
    Content-Length: 909

    v=0
    o=user 1074707710 1074707710 IN IP4 192.168.1.222
    s=Asterisk PBX 11.0.1
    c=IN IP4 192.168.1.222
    t=0 0
    m=audio 14672 RTP/AVP 10 4 3 3 0 0 8 8 112 5 7 18 110 97 111 9 102 115 116 117 118 119
    a=rtpmap:10 L16/8000
    a=rtpmap:4 G723/8000
    a=fmtp:4 annexa=no
    a=rtpmap:3 GSM/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:112 AAL2-G726-32/8000
    a=rtpmap:5 DVI4/8000
    a=rtpmap:7 LPC/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:110 speex/8000
    a=rtpmap:97 iLBC/8000
    a=fmtp:97 mode=30
    a=rtpmap:111 G726-32/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:102 G7221/16000
    a=fmtp:102 bitrate=32000
    a=rtpmap:115 G7221/32000
    a=fmtp:115 bitrate=48000
    a=rtpmap:116 G719/48000
    a=fmtp:116 bitrate=64000
    a=rtpmap:117 speex/16000
    a=rtpmap:118 L16/16000
    a=rtpmap:119 speex/32000
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv

    ---
    [Feb  7 17:40:48] VERBOSE[7572] chan_sip.c: 
    <--- SIP read from UDP:193.201.229.35:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.222:5060;received=83.246.243.5;branch=z9hG4bK1cb00090;rport=5060
    From: "7929*******" <sip:7929*******@multifon.ru>;tag=as59f41166
    To: <sip:79299999999@193.201.229.35:5060>
    Call-ID: 32c8275544b26a577d38a7ad072a4ef6@multifon.ru
    CSeq: 102 INVITE

    <------------->
    [Feb  7 17:40:48] VERBOSE[7572] chan_sip.c: --- (6 headers 0 lines) ---
    [Feb  7 17:40:48] VERBOSE[7572] chan_sip.c: 
    <--- SIP read from UDP:193.201.229.35:5060 --->
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 192.168.1.222:5060;received=83.246.243.5;branch=z9hG4bK1cb00090;rport=5060
    From: "7929*******" <sip:7929*******@multifon.ru>;tag=as59f41166
    To: <sip:79299999999@193.201.229.35:5060>;tag=A55D3246313536414C3A3D00
    Call-ID: 32c8275544b26a577d38a7ad072a4ef6@multifon.ru
    CSeq: 102 INVITE
    Proxy-Authenticate: Digest nonce="MTM2MDIzMzcxNjqXn8k0xr7Pa+IewmFHjDjb",opaque="MTM2MDIzMzcxNjqXn8k0xr7Pa+IewmFHjDjb",algorithm=md5,realm="BREDBAND",qop="auth"
    Content-Length: 0

    <------------->
    [Feb  7 17:40:48] VERBOSE[7572] chan_sip.c: --- (8 headers 0 lines) ---
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Transmitting (NAT) to 193.201.229.35:5060:
    ACK sip:79299999999@193.201.229.35:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.222:5060;branch=z9hG4bK1cb00090;rport
    Max-Forwards: 70
    From: "7929*******" <sip:7929*******@multifon.ru>;tag=as59f41166
    To: <sip:79299999999@193.201.229.35:5060>;tag=A55D3246313536414C3A3D00
    Contact: <sip:7929*******@192.168.1.222:5060>
    Call-ID: 32c8275544b26a577d38a7ad072a4ef6@multifon.ru
    CSeq: 102 ACK
    User-Agent: Asterisk PBX 11.0.1
    Content-Length: 0


    ---
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: We think we can do text
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Audio is at 14672
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100019 (slin) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100001 (g723) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100002 (gsm) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100002 (gsm) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100003 (ulaw) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100003 (ulaw) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100004 (alaw) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100004 (alaw) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100005 (g726aal2) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100006 (adpcm) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100007 (lpc10) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100008 (g729) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100009 (speex) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100010 (ilbc) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100011 (g726) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100012 (g722) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100013 (siren7) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100014 (siren14) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100015 (g719) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100016 (speex16) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100017 (testlaw) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100017 (testlaw) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100020 (slin12) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100021 (slin16) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100022 (slin24) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100023 (slin32) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100024 (slin44) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100025 (slin48) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100026 (slin96) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100027 (slin192) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Adding codec 100028 (speex32) to SDP
    [Feb  7 17:40:48] VERBOSE[7572][C-00000023] chan_sip.c: Reliably Transmitting (NAT) to 193.201.229.35:5060:
    INVITE sip:79299999999@193.201.229.35:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.222:5060;branch=z9hG4bK6b13e2af;rport
    Max-Forwards: 70
    From: "7929*******" <sip:7929*******@multifon.ru>;tag=as59f41166
    To: <sip:79299999999@193.201.229.35:5060>
    Contact: <sip:7929*******@192.168.1.222:5060>
    Call-ID: 32c8275544b26a577d38a7ad072a4ef6@multifon.ru
    CSeq: 103 INVITE
    User-Agent: Asterisk PBX 11.0.1
    Proxy-Authorization: Digest username="7929*******", realm="BREDBAND", algorithm=MD5, uri="sip:79299999999@193.201.229.35:5060", nonce="MTM2MDIzMzcxNjqXn8k0xr7Pa+IewmFHjDjb", response="80623c42ef00b67e3df075271e131393", opaque="MTM2MDIzMzcxNjqXn8k0xr7Pa+IewmFHjDjb", qop=auth, cnonce="50cdfe78", nc=00000001
    Date: Thu, 07 Feb 2013 10:40:48 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Remote-Party-ID: "7929*******" <sip:7929*******@multifon.ru>;party=calling;privacy=off;screen=no
    Content-Type: application/sdp
    Content-Length: 909

    v=0
    o=user 1074707710 1074707711 IN IP4 192.168.1.222
    s=Asterisk PBX 11.0.1
    c=IN IP4 192.168.1.222
    t=0 0
    m=audio 14672 RTP/AVP 10 4 3 3 0 0 8 8 112 5 7 18 110 97 111 9 102 115 116 117 118 119
    a=rtpmap:10 L16/8000
    a=rtpmap:4 G723/8000
    a=fmtp:4 annexa=no
    a=rtpmap:3 GSM/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:112 AAL2-G726-32/8000
    a=rtpmap:5 DVI4/8000
    a=rtpmap:7 LPC/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:110 speex/8000
    a=rtpmap:97 iLBC/8000
    a=fmtp:97 mode=30
    a=rtpmap:111 G726-32/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:102 G7221/16000
    a=fmtp:102 bitrate=32000
    a=rtpmap:115 G7221/32000
    a=fmtp:115 bitrate=48000
    a=rtpmap:116 G719/48000
    a=fmtp:116 bitrate=64000
    a=rtpmap:117 speex/16000
    a=rtpmap:118 L16/16000
    a=rtpmap:119 speex/32000
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv

    ---
    [Feb  7 17:40:48] VERBOSE[7572] chan_sip.c: 
    <--- SIP read from UDP:193.201.229.35:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.222:5060;received=83.246.243.5;branch=z9hG4bK6b13e2af;rport=5060
    From: "7929*******" <sip:7929*******@multifon.ru>;tag=as59f41166
    To: <sip:79299999999@193.201.229.35:5060>
    Call-ID: 32c8275544b26a577d38a7ad072a4ef6@multifon.ru
    CSeq: 103 INVITE

    <------------->
    [Feb  7 17:40:48] VERBOSE[7572] chan_sip.c: --- (6 headers 0 lines) ---
    [Feb  7 17:40:49] VERBOSE[7572] chan_sip.c: 
    <--- SIP read from UDP:193.201.229.35:5060 --->
    SIP/2.0 488 Not Acceptable Here
    Via: SIP/2.0/UDP 192.168.1.222:5060;received=83.246.243.5;branch=z9hG4bK6b13e2af;rport=5060
    From: "7929*******" <sip:7929*******@multifon.ru>;tag=as59f41166
    To: <sip:79299999999@193.201.229.35:5060>;tag=E788324631353641533A3D00
    Call-ID: 32c8275544b26a577d38a7ad072a4ef6@multifon.ru
    CSeq: 103 INVITE
    Content-Length: 0

    <------------->
    [Feb  7 17:40:49] VERBOSE[7572] chan_sip.c: --- (7 headers 0 lines) ---
    [Feb  7 17:40:49] VERBOSE[7572][C-00000023] chan_sip.c: Transmitting (NAT) to 193.201.229.35:5060:
    ACK sip:79299999999@193.201.229.35:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.222:5060;branch=z9hG4bK6b13e2af;rport
    Max-Forwards: 70
    From: "7929*******" <sip:7929*******@multifon.ru>;tag=as59f41166
    To: <sip:79299999999@193.201.229.35:5060>;tag=E788324631353641533A3D00
    Contact: <sip:7929*******@192.168.1.222:5060>
    Call-ID: 32c8275544b26a577d38a7ad072a4ef6@multifon.ru
    CSeq: 103 ACK
    User-Agent: Asterisk PBX 11.0.1
    Content-Length: 0


    ---
    [Feb  7 17:40:49] VERBOSE[7671][C-00000023] chan_sip.c: Scheduling destruction of SIP dialog '32c8275544b26a577d38a7ad072a4ef6@multifon.ru' in 32000 ms (Method: INVITE)
    [Feb  7 17:40:49] NOTICE[7671] pbx_spool.c: Call failed to go through, reason (1) Hangup
    [Feb  7 17:40:49] NOTICE[7671] pbx_spool.c: Queued call to SIP/79299999999@multifon-out expired without completion after 0 attempts
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ответил 2013-02-07 14:53:10 +0400

tlq Gravatar tlq
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Задан: 2013-02-07 13:05:11 +0400

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Обновлен: Feb 07 '13

Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.