![]() | 1 | изначальная версия редактировать | |
Добрый день, Подскажите что нет так? Мучусь уже 2 день.
[internal] exten => XXX,1,Dial(SIP/${EXTEN}) exten => XXX,n, Hangupexten => _7925XXXXXXX,1,Dial(SIP/multifon/${EXTEN},120) exten => _7925XXXXXXX,n,HangUp
[from-multifon] exten => _x.,1,GotoIf($["${CALLERID(num)}" = "7925xxxxxxx"]?callback,s,1) exten => _x.,n,Hangup
[callback] exten => s,1,System(/etc/asterisk/scripts/callback ${CALLERID(num)} &) exten => s,n,Hangup
[disa1] exten => s,1,Answer exten => s,n,Wait(2) exten => s,n,Playback(please-enter-your) exten => s,n,Background(telephone-number) exten => s,n,Set(TIMEOUT(digit)=5) exten => s,n,Set(TIMEOUT(response)=10) exten => s,n,DISA(no-password,default) exten => s,n,HangUp
Содержимое /etc/asterisk/scripts/callback
#!/bin/sh sleep 15 NUMBER=$1 echo "Channel: SIP/internal/$NUMBER MaxRetries: 1 RetryTime: 10 WaitTime: 20 Context: disa1 Extension: 222 Priority: 1 AlwaysDelete: Yes" >/var/spool/asterisk/tmp/$NUMBER mv /var/spool/asterisk/tmp/$NUMBER /var/spool/asterisk/outgoing/$NUMBERВот что получается
== Using SIP RTP CoS mark 5 -- Executing [792yyyyyyy@from-multifon:1] GotoIf("SIP/multifon-0000001b", "1?callback,s,1") in new stack -- Goto (callback,s,1) -- Executing [s@callback:1] System("SIP/multifon-0000001b", "/etc/asterisk/scripts/callback 7925yyyyyyy &") in new stack -- Executing [s@callback:2] Hangup("SIP/multifon-0000001b", "") in new stack == Spawn extension (callback, s, 2) exited non-zero on 'SIP/multifon-0000001b'
![]() | 2 | No.2 Revision редактировать |
Добрый день, Подскажите что нет так? Мучусь уже 2 день.
[internal] exten => XXX,1,Dial(SIP/${EXTEN}) exten => XXX,n, Hangupexten => _7925XXXXXXX,1,Dial(SIP/multifon/${EXTEN},120) exten => _7925XXXXXXX,n,HangUp
[from-multifon] exten => _x.,1,GotoIf($["${CALLERID(num)}" = "7925xxxxxxx"]?callback,s,1) exten => _x.,n,Hangup
[callback] exten => s,1,System(/etc/asterisk/scripts/callback ${CALLERID(num)} &) exten => s,n,Hangup
[disa1] exten => s,1,Answer exten => s,n,Wait(2) exten => s,n,Playback(please-enter-your) exten => s,n,Background(telephone-number) exten => s,n,Set(TIMEOUT(digit)=5) exten => s,n,Set(TIMEOUT(response)=10) exten => s,n,DISA(no-password,default) exten => s,n,HangUp
Содержимое /etc/asterisk/scripts/callback
#!/bin/sh sleep 15 NUMBER=$1 echo "Channel: SIP/internal/$NUMBER MaxRetries: 1 RetryTime: 10 WaitTime: 20 Context: disa1 Extension: 222 Priority: 1 AlwaysDelete: Yes" >/var/spool/asterisk/tmp/$NUMBER mv /var/spool/asterisk/tmp/$NUMBER /var/spool/asterisk/outgoing/$NUMBERВот что получается
== Using SIP RTP CoS mark 5 -- Executing [792yyyyyyy@from-multifon:1] GotoIf("SIP/multifon-0000001b", "1?callback,s,1") in new stack -- Goto (callback,s,1) -- Executing [s@callback:1] System("SIP/multifon-0000001b", "/etc/asterisk/scripts/callback 7925yyyyyyy &") in new stack -- Executing [s@callback:2] Hangup("SIP/multifon-0000001b", "") in new stack == Spawn extension (callback, s, 2) exited non-zero on 'SIP/multifon-0000001b'
Не поверите, до сих пор не смог победить колбек с дисой. Ситуация такая колбек и disa работает (немного перемудрил с первичным диалпланом), но только в режиме подкладки call файла вручную. Тескт файла Channel: SIP/multifon/79251234567 MaxRetries: 3 RetryTime: 60 WaitTime: 30 Context: disa1 Extension: s Archive: Yes
Но скрипт этот в сочетании с екстеншином exten => s,1,System(/etc/asterisk/scripts/callback ${CALLERID(num)} > /tmp/callbackerror 2>&1 &) exten => s,n,Hangup
#!/bin/sh sleep 20 echo "Channel: SIP/multifon/$NUMBER MaxRetries: 3 RetryTime: 60 WaitTime: 30 Context: disa1 Extension: s Archive: Yes" >/var/spool/asterisk/tmp/$NUMBER mv /var/spool/asterisk/tmp/$NUMBER /var/spool/asterisk/outgoing/$NUMBER
Не дает мне нужного результата. Мозги кипят, что делать. У меня подозрение, что он не передает номер вызывающего абонента в call файл.
![]() | 3 | No.3 Revision редактировать |
Добрый день,
Подскажите что нет так? Мучусь уже 2 день.
[internal] exten => XXX,1,Dial(SIP/${EXTEN}) exten => XXX,n,HangupHangup exten => _7925XXXXXXX,1,Dial(SIP/multifon/${EXTEN},120) exten =>_7925XXXXXXX,n,HangUp_7925XXXXXXX,n,HangUp [from-multifon] exten => _x.,1,GotoIf($["${CALLERID(num)}" = "7925xxxxxxx"]?callback,s,1) exten =>_x.,n,Hangup_x.,n,Hangup [callback] exten => s,1,System(/etc/asterisk/scripts/callback ${CALLERID(num)} &) exten =>s,n,Hangups,n,Hangup [disa1] exten => s,1,Answer exten => s,n,Wait(2) exten => s,n,Playback(please-enter-your) exten => s,n,Background(telephone-number) exten => s,n,Set(TIMEOUT(digit)=5) exten => s,n,Set(TIMEOUT(response)=10) exten => s,n,DISA(no-password,default) exten => s,n,HangUp
Содержимое /etc/asterisk/scripts/callback
#!/bin/sh sleep 15 NUMBER=$1 echo "Channel: SIP/internal/$NUMBER MaxRetries: 1 RetryTime: 10 WaitTime: 20 Context: disa1 Extension: 222 Priority: 1 AlwaysDelete: Yes" >/var/spool/asterisk/tmp/$NUMBER mv /var/spool/asterisk/tmp/$NUMBER /var/spool/asterisk/outgoing/$NUMBERВот что получается
== Using SIP RTP CoS mark 5 -- Executing [792yyyyyyy@from-multifon:1] GotoIf("SIP/multifon-0000001b", "1?callback,s,1") in new stack -- Goto (callback,s,1) -- Executing [s@callback:1] System("SIP/multifon-0000001b", "/etc/asterisk/scripts/callback 7925yyyyyyy &") in new stack -- Executing [s@callback:2] Hangup("SIP/multifon-0000001b", "") in new stack == Spawn extension (callback, s, 2) exited non-zero on 'SIP/multifon-0000001b'
Не поверите, до сих пор не смог победить колбек с дисой. Ситуация такая колбек и disa работает (немного перемудрил с первичным диалпланом), но только в режиме подкладки call файла вручную. Тескт файла Channel: SIP/multifon/79251234567 MaxRetries: 3 RetryTime: 60 WaitTime: 30 Context: disa1 Extension: s Archive:YesYes Но скрипт этот в сочетании с екстеншином exten => s,1,System(/etc/asterisk/scripts/callback ${CALLERID(num)} > /tmp/callbackerror 2>&1 &) exten =>s,n,Hangups,n,Hangup #!/bin/sh sleep 20 echo "Channel: SIP/multifon/$NUMBER MaxRetries: 3 RetryTime: 60 WaitTime: 30 Context: disa1 Extension: s Archive: Yes" >/var/spool/asterisk/tmp/$NUMBER mv /var/spool/asterisk/tmp/$NUMBER /var/spool/asterisk/outgoing/$NUMBERНе дает мне нужного результата. Мозги кипят, что делать. У меня подозрение, что он не передает номер вызывающего абонента в callфайл.файл.<------------> -- Executing [79261945604@from-multifon:1] GotoIf("SIP/multifon-00000002", "1?callback1,s,1") in new stack -- Goto (callback1,s,1) -- Executing [s@callback1:1] System("SIP/multifon-00000002", "/etc/asterisk/scripts/callback-1 &") in new stack -- Executing [s@callback1:2] Hangup("SIP/multifon-00000002", "") in new stack
== Spawn extension (callback1, s, 2) exited non-zero on 'SIP/multifon-00000002' Scheduling destruction of SIP dialog '020231880581400000007914@SFESIP1-id1-ext' in 6400 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 193.201.229.35:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK5afuf01008v00p89r7s1.1;received=193.201.229.35;rport=5060 From: <sip:79251891044@10.190.35.17>;tag=95ffcd055e0f78f7d5d397020e89288d0725de9f To: sip:79261945604-qr4vc4rvrgl30@10.190.35.4:5060;tag=as23833ad9 Call-ID: 020231880581400000007914@SFESIP1-id1-ext CSeq: 1 INVITE Server: Asterisk PBX 1.8.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0
<------------>
<--- SIP read from UDP:193.201.229.35:5060 ---> ACK sip:79261945604@37.59.239.64:5060 SIP/2.0 Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK5afuf01008v00p89r7s1.1 CSeq: 1 ACK Max-Forwards: 19 From: <sip:79251891044@10.190.35.17>;tag=95ffcd055e0f78f7d5d397020e89288d0725de9f To: <sip:79261945604-qr4vc4rvrgl30@10.190.35.4:5060>;tag=as23833ad9 Call-ID: 020231880581400000007914@SFESIP1-id1-ext Content-Length: 0
<-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '79f7c47b0195ccf046b6a13007c96624@37.59.239.64' Method: REGISTER Really destroying SIP dialog '020231880581400000007914@SFESIP1-id1-ext' Method: ACK
<--- SIP read from UDP:178.63.16.146:5060 ---> SIP/2.0 200 OK CSeq: 164 REGISTER Via: SIP/2.0/UDP 37.59.239.64:5060;branch=z9hG4bK0b8d5e22;rport From: <sip:1000054983@soft.mob1.biz>;tag=as0e8a3418 Call-ID: 6db054283765f29f4619f6fb348ae9a8@37.59.239.64 To: <sip:1000054983@soft.mob1.biz>;tag=1150582588654835 Contact: <sip:1000054983@37.59.239.64:5060>;expires=40 Expires: 40 Content-Length: 0
<-------------> --- (9 headers 0 lines) --- Scheduling destruction of SIP dialog '6db054283765f29f4619f6fb348ae9a8@37.59.239.64' in 32000 ms (Method: REGISTER) Reliably Transmitting (NAT) to 213.133.98.3:5060: OPTIONS sip:222@213.133.98.3 SIP/2.0 Via: SIP/2.0/UDP 37.59.239.64:5060;branch=z9hG4bK04bde992;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@37.59.239.64>;tag=as6a6d78f4 To: <sip:222@213.133.98.3> Contact: <sip:asterisk@37.59.239.64:5060> Call-ID: 1fc30bd141e953f40bc7ec3973f5f076@37.59.239.64:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.12.0 Date: Fri, 03 Aug 2012 05:50:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0
<--- SIP read from UDP:213.133.98.3:5060 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 37.59.239.64:5060;branch=z9hG4bK04bde992;received=37.59.239.64;rport=5060 From: "asterisk" <sip:asterisk@37.59.239.64>;tag=as6a6d78f4 To: <sip:222@213.133.98.3> Call-ID: 1fc30bd141e953f40bc7ec3973f5f076@37.59.239.64:5060 CSeq: 102 OPTIONS Content-Length: 0
<-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '1fc30bd141e953f40bc7ec3973f5f076@37.59.239.64:5060' Method: OPTIONS Reliably Transmitting (NAT) to 193.201.229.35:5060: OPTIONS sip:multifon.ru SIP/2.0 Via: SIP/2.0/UDP 37.59.239.64:5060;branch=z9hG4bK68689680;rport Max-Forwards: 70 From: "asterisk" <sip:79261945604@37.59.239.64>;tag=as43eeadac To: <sip:multifon.ru> Contact: <sip:79261945604@37.59.239.64:5060> Call-ID: 27d6278057fd06c8390048807e4db2a3@37.59.239.64:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.12.0 Date: Fri, 03 Aug 2012 05:50:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0
<--- SIP read from UDP:193.201.229.35:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 37.59.239.64:5060;received=37.59.239.64;branch=z9hG4bK68689680;rport=5060 From: "asterisk" <sip:79261945604@37.59.239.64>;tag=as43eeadac To: <sip:multifon.ru>;tag=aprqngfrt-1n7qa230000c6 Call-ID: 27d6278057fd06c8390048807e4db2a3@37.59.239.64:5060 CSeq: 102 OPTIONS Reason: Q.850;cause=55;text="Call Terminated"
<-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '27d6278057fd06c8390048807e4db2a3@37.59.239.64:5060' Method: OPTIONS Really destroying SIP dialog '6db054283765f29f4619f6fb348ae9a8@37.59.239.64' Method: REGISTER Really destroying SIP dialog '61cd176b77802aad0034aff938dab2fb@37.59.239.64' Method: REGISTER i-peak*CLI>
![]() | 4 | No.4 Revision редактировать |
Добрый день, Подскажите что нет так? Мучусь уже 2 день.
[internal]
exten => XXX,1,Dial(SIP/${EXTEN})
exten => XXX,n, Hangup
exten => _7925XXXXXXX,1,Dial(SIP/multifon/${EXTEN},120)
exten => _7925XXXXXXX,n,HangUp
[from-multifon]
exten => _x.,1,GotoIf($["${CALLERID(num)}" = "7925xxxxxxx"]?callback,s,1)
exten => _x.,n,Hangup
[callback]
exten => s,1,System(/etc/asterisk/scripts/callback ${CALLERID(num)} &)
exten => s,n,Hangup
[disa1]
exten => s,1,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(please-enter-your)
exten => s,n,Background(telephone-number)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,DISA(no-password,default)
exten => s,n,HangUp
Содержимое /etc/asterisk/scripts/callback
#!/bin/sh
sleep 15
NUMBER=$1
echo "Channel: SIP/internal/$NUMBER
MaxRetries: 1
RetryTime: 10
WaitTime: 20
Context: disa1
Extension: 222
Priority: 1
AlwaysDelete: Yes" >/var/spool/asterisk/tmp/$NUMBER
mv /var/spool/asterisk/tmp/$NUMBER /var/spool/asterisk/outgoing/$NUMBER
Вот что получается
== Using SIP RTP CoS mark 5
-- Executing [792yyyyyyy@from-multifon:1] GotoIf("SIP/multifon-0000001b", "1?callback,s,1") in new stack
-- Goto (callback,s,1)
-- Executing [s@callback:1] System("SIP/multifon-0000001b", "/etc/asterisk/scripts/callback 7925yyyyyyy &") in new stack
-- Executing [s@callback:2] Hangup("SIP/multifon-0000001b", "") in new stack
== Spawn extension (callback, s, 2) exited non-zero on 'SIP/multifon-0000001b'
Не поверите, до сих пор не смог победить колбек с дисой.
Ситуация такая колбек и disa работает (немного перемудрил с первичным диалпланом), но только в режиме подкладки call файла вручную.
Тескт файла
Channel: SIP/multifon/79251234567
MaxRetries: 3
RetryTime: 60
WaitTime: 30
Context: disa1
Extension: s
Archive: Yes
Но скрипт этот в сочетании с екстеншином exten => s,1,System(/etc/asterisk/scripts/callback ${CALLERID(num)} > /tmp/callbackerror 2>&1 &)
exten => s,n,Hangup
#!/bin/sh
sleep 20
echo "Channel: SIP/multifon/$NUMBER
MaxRetries: 3
RetryTime: 60
WaitTime: 30
Context: disa1
Extension: s
Archive: Yes" >/var/spool/asterisk/tmp/$NUMBER
mv /var/spool/asterisk/tmp/$NUMBER /var/spool/asterisk/outgoing/$NUMBER
Не дает мне нужного результата. Мозги кипят, что делать. У меня подозрение, что он не передает номер вызывающего абонента в call файл.
<------------>
-- Executing [79261945604@from-multifon:1] GotoIf("SIP/multifon-00000002", "1?callback1,s,1") in new stack
-- Goto (callback1,s,1)
-- Executing [s@callback1:1] System("SIP/multifon-00000002", "/etc/asterisk/scripts/callback-1 &") in new stack
-- Executing [s@callback1:2] Hangup("SIP/multifon-00000002", "") in new stack
== Spawn extension (callback1, s, 2) exited non-zero on 'SIP/multifon-00000002'
Scheduling destruction of SIP dialog '020231880581400000007914@SFESIP1-id1-ext' in 6400 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 193.201.229.35:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK5afuf01008v00p89r7s1.1;received=193.201.229.35;rport=5060
From: <sip:79251891044@10.190.35.17>;tag=95ffcd055e0f78f7d5d397020e89288d0725de9f
To: sip:79261945604-qr4vc4rvrgl30@10.190.35.4:5060;tag=as23833ad9
Call-ID: 020231880581400000007914@SFESIP1-id1-ext
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:193.201.229.35:5060 --->
ACK sip:79261945604@37.59.239.64:5060 SIP/2.0
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK5afuf01008v00p89r7s1.1
CSeq: 1 ACK
Max-Forwards: 19
From: <sip:79251891044@10.190.35.17>;tag=95ffcd055e0f78f7d5d397020e89288d0725de9f
To: <sip:79261945604-qr4vc4rvrgl30@10.190.35.4:5060>;tag=as23833ad9
Call-ID: 020231880581400000007914@SFESIP1-id1-ext
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '79f7c47b0195ccf046b6a13007c96624@37.59.239.64' Method: REGISTER
Really destroying SIP dialog '020231880581400000007914@SFESIP1-id1-ext' Method: ACK
<--- SIP read from UDP:178.63.16.146:5060 --->
SIP/2.0 200 OK
CSeq: 164 REGISTER
Via: SIP/2.0/UDP 37.59.239.64:5060;branch=z9hG4bK0b8d5e22;rport
From: <sip:1000054983@soft.mob1.biz>;tag=as0e8a3418
Call-ID: 6db054283765f29f4619f6fb348ae9a8@37.59.239.64
To: <sip:1000054983@soft.mob1.biz>;tag=1150582588654835
Contact: <sip:1000054983@37.59.239.64:5060>;expires=40
Expires: 40
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '6db054283765f29f4619f6fb348ae9a8@37.59.239.64' in 32000 ms (Method: REGISTER)
Reliably Transmitting (NAT) to 213.133.98.3:5060:
OPTIONS sip:222@213.133.98.3 SIP/2.0
Via: SIP/2.0/UDP 37.59.239.64:5060;branch=z9hG4bK04bde992;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@37.59.239.64>;tag=as6a6d78f4
To: <sip:222@213.133.98.3>
Contact: <sip:asterisk@37.59.239.64:5060>
Call-ID: 1fc30bd141e953f40bc7ec3973f5f076@37.59.239.64:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.12.0
Date: Fri, 03 Aug 2012 05:50:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<--- SIP read from UDP:213.133.98.3:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 37.59.239.64:5060;branch=z9hG4bK04bde992;received=37.59.239.64;rport=5060
From: "asterisk" <sip:asterisk@37.59.239.64>;tag=as6a6d78f4
To: <sip:222@213.133.98.3>
Call-ID: 1fc30bd141e953f40bc7ec3973f5f076@37.59.239.64:5060
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '1fc30bd141e953f40bc7ec3973f5f076@37.59.239.64:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 193.201.229.35:5060:
OPTIONS sip:multifon.ru SIP/2.0
Via: SIP/2.0/UDP 37.59.239.64:5060;branch=z9hG4bK68689680;rport
Max-Forwards: 70
From: "asterisk" <sip:79261945604@37.59.239.64>;tag=as43eeadac
To: <sip:multifon.ru>
Contact: <sip:79261945604@37.59.239.64:5060>
Call-ID: 27d6278057fd06c8390048807e4db2a3@37.59.239.64:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.12.0
Date: Fri, 03 Aug 2012 05:50:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 37.59.239.64:5060;received=37.59.239.64;branch=z9hG4bK68689680;rport=5060
From: "asterisk" <sip:79261945604@37.59.239.64>;tag=as43eeadac
To: <sip:multifon.ru>;tag=aprqngfrt-1n7qa230000c6
Call-ID: 27d6278057fd06c8390048807e4db2a3@37.59.239.64:5060
CSeq: 102 OPTIONS
Reason: Q.850;cause=55;text="Call Terminated"
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '27d6278057fd06c8390048807e4db2a3@37.59.239.64:5060' Method: OPTIONS
Really destroying SIP dialog '6db054283765f29f4619f6fb348ae9a8@37.59.239.64' Method: REGISTER
Really destroying SIP dialog '61cd176b77802aad0034aff938dab2fb@37.59.239.64' Method: REGISTER
i-peak*CLI>
Проблема была в скрипте!
Вот рабочий вариант
#!/bin/sh
sleep 5
NUMBER=$1
echo "Channel: SIP/multifon/$NUMBER
MaxRetries: 1
RetryTime: 30
WaitTime: 30
Context: disa1
Extension: s
Priority: 1
AlwaysDelete: Yes
" > /var/spool/asterisk/tmp/$NUMBER.call
mv /var/spool/asterisk/tmp/$NUMBER.call /var/spool/asterisk/outgoing/
Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании
Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией
GNU GPL.