[AsteriskBrasil] DISA e Interligacao entre asterisk (via SIP)
Itamar Reis Peixoto
itamar em ispbrasil.com.br
Sexta Julho 10 10:07:11 BRT 2009
se quiser que eu te explique uma maneira facil de fazer isto é só me
ligar no sip: itamar em ispbrasil.com.br
vc pode baixar o ekiga pra windows e colocar o meu e-mail na caixinha
e mandar discar.
2009/7/10 César Davi Avila do Nascimento <cesargxn em gmail.com>:
>
> Pessoal,
>
> Estou precisando testar o seguinte cenário:
>
> +-----------+ +-----------+
> | asterisk 1| | asterisk 2|
> +-----------+ +-----------+
>
> | |
>
> | |
> _______|__________________|___________
> | |
> | |
> | |
> +-------+ +-------+
>
>
> | ATA 1 | | ATA 2 |
> +-------+ +-------+
> / \ / \
> / \ / \
>
> 21 22 10 11
>
>
> Ou seja, tenho 2 asterisks interligados via SIP, dois ATAs com duas linhas,
> sendo que o ATA1 está registrado no asterisk 1 e o ATA 2 está registrado no
> asterisk 2 e, todas as chamadas entrantes no asterisk2 vindas do asterisk 1
> (via SIP), são atendidas por um DISA.
>
> Consigo fazer ligações do ATA 1 para o ATA 2 sem problemas (a chamada vai
> até o asterisk1 é roteada para o asterisk 2, cai no DISA e eu chamo um dos
> telefones do ATA2). Estou tentando agora fazer com que a chamada vinda por
> exemplo do ramal 21, vá até o asterisk 2, caia no DISA e retorne para o
> asterisk 1 (no ramal 22).
>
>
>
> Como sou newbie no assunto, gostaria de saber com os amigos da lista se isto
> é possível... Ou se existe uma outra forma de fazer isso....
> Abaixo segue meus arquivos de conf.
>
> Grande abraço
>
> César
>
>
>
> ===============================================================================================================================
>
> Arquivos de conf do asterisk 1
>
> ******
> sip.conf
> ********
>
> [21]
> type=friend
>
>
> context=phones ; Where to start in the dialplan when this
> phone calls
> secret=21
> ;callerid=John Doe <1234> ; Full caller ID, to override the phones
> config
> ; on incoming calls to Asterisk
>
>
> host=dynamic ; we have a static but private IP address
> ; No registration allowed
> ;nat=no ; there is not NAT between phone and
> Asterisk
> ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
>
>
> ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
> ;call-limit=1 ; permit only 1 outgoing call and 1 incoming
> call at a time
> ; from the phone to asterisk
>
>
> ; 1 for the explicit peer, 1 for the
> explicit user,
> ; remember that a friend equals 1 peer and 1
> user in
> ; memory
> ; This will affect your subscriptions as
> well.
>
>
> ; There is no combined call counter for a
> "friend"
> ; so there's currently no way in sip.conf to
> limit
> ; to one inbound or outbound call per phone.
> Use
>
>
> ; the group counters in the dial plan for
> that.
> ;
> ;mailbox=1234 em default ; mailbox 1234 in voicemail context
> "default"
> disallow=all ; need to disallow=all before we can use
> allow=
>
>
> allow=ulaw ; Note: In user sections the order of codecs
> ; listed with allow= does NOT matter!
> allow=alaw
> allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
>
>
> allow=g729 ; Pass-thru only unless g729 license obtained
> ;callingpres=allowed_passed_screen ; Set caller ID presentation
> ; See doc/callingpres.txt for more
> information
>
>
>
> [22]
> type=friend
> context=phones ; Where to start in the dialplan when this
> phone calls
> secret=22
> ;callerid=John Doe <1234> ; Full caller ID, to override the phones
> config
>
>
> ; on incoming calls to Asterisk
> host=dynamic ; we have a static but private IP address
> ; No registration allowed
> ;nat=no ; there is not NAT between phone and
> Asterisk
>
>
> ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
> ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
> ;call-limit=1 ; permit only 1 outgoing call and 1 incoming
> call at a time
>
>
> ; from the phone to asterisk
> ; 1 for the explicit peer, 1 for the
> explicit user,
> ; remember that a friend equals 1 peer and 1
> user in
>
>
> ; memory
> ; This will affect your subscriptions as
> well.
> ; There is no combined call counter for a
> "friend"
>
>
> ; so there's currently no way in sip.conf to
> limit
> ; to one inbound or outbound call per phone.
> Use
> ; the group counters in the dial plan for
> that.
>
>
> ;
> ;mailbox=1234 em default ; mailbox 1234 in voicemail context
> "default"
> disallow=all ; need to disallow=all before we can use
> allow=
> allow=ulaw ; Note: In user sections the order of codecs
>
>
> ; listed with allow= does NOT matter!
> allow=alaw
> allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
> allow=g729 ; Pass-thru only unless g729 license obtained
>
>
> ;callingpres=allowed_passed_screen ; Set caller ID presentation
> ; See doc/callingpres.txt for more
> information
>
> ;comunicação entre asterisks
>
> [asterisk2]
> type=friend
>
>
> secret=welcome
> context=asterisk2_incoming
> host=dynamic
> disallow=all ; need to disallow=all before we can use
> allow=
> allow=ulaw ; Note: In user sections the order of codecs
>
>
> ; listed with allow= does NOT matter!
> allow=alaw
> allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
> allow=g729 ; Pass-thru only unless g729 license obtained
>
>
>
> ******
> extensions.conf
> ******
>
> [phones]
> include=>internal
> include=>remote
>
>
> [internal]
> exten=>_2x,1,NoOp()
> exten=>_2x,n,Dial(SIP/${EXTEN},30)
> exten=>_2x,n,Hangup()
>
>
>
> [remote]
> ;exten=>_1x,1,NoOp()
> exten=>_1x,1,Dial(SIP/asterisk2/${EXTEN})
> exten=>_3x,1,Dial(SIP/asterisk2/${EXTEN})
> exten=>_1x,n+101,Hangup()
> exten=>_3x,n+101,Hangup()
>
> [asterisk2_incoming]
>
>
> include=>internal
>
> **************************************************
> Arquivos de conf do asterisk 2
>
> ******
> sip.conf
> *******
>
> [10]
> type=friend
> context=phones ; Where to start in the dialplan when this
> phone calls
>
>
> secret=10
> ;callerid=John Doe <1234> ; Full caller ID, to override the phones
> config
> ; on incoming calls to Asterisk
> host=dynamic ; we have a static but private IP address
>
>
> ; No registration allowed
> ;nat=no ; there is not NAT between phone and
> Asterisk
> ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
>
>
> ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
> ;call-limit=1 ; permit only 1 outgoing call and 1 incoming
> call at a time
> ; from the phone to asterisk
>
>
> ; 1 for the explicit peer, 1 for the
> explicit user,
> ; remember that a friend equals 1 peer and 1
> user in
> ; memory
> ; This will affect your subscriptions as
> well.
>
>
> ; There is no combined call counter for a
> "friend"
> ; so there's currently no way in sip.conf to
> limit
> ; to one inbound or outbound call per phone.
> Use
>
>
> ; the group counters in the dial plan for
> that.
> ;
> ;mailbox=1234 em default ; mailbox 1234 in voicemail context
> "default"
> disallow=all ; need to disallow=all before we can use
> allow=
>
>
> allow=ulaw ; Note: In user sections the order of codecs
> ; listed with allow= does NOT matter!
> allow=alaw
> allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
>
>
> allow=g729 ; Pass-thru only unless g729 license obtained
> ;callingpres=allowed_passed_screen ; Set caller ID presentation
> ; See doc/callingpres.txt for more
> information
>
>
>
> [11]
> type=friend
> context=phones ; Where to start in the dialplan when this
> phone calls
> secret=11
> ;callerid=John Doe <1234> ; Full caller ID, to override the phones
> config
>
>
> ; on incoming calls to Asterisk
> host=dynamic ; we have a static but private IP address
> ; No registration allowed
> ;nat=no ; there is not NAT between phone and
> Asterisk
>
>
> ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
> ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
> ;call-limit=1 ; permit only 1 outgoing call and 1 incoming
> call at a time
>
>
> ; from the phone to asterisk
> ; 1 for the explicit peer, 1 for the
> explicit user,
> ; remember that a friend equals 1 peer and 1
> user in
>
>
> ; memory
> ; This will affect your subscriptions as
> well.
> ; There is no combined call counter for a
> "friend"
>
>
> ; so there's currently no way in sip.conf to
> limit
> ; to one inbound or outbound call per phone.
> Use
> ; the group counters in the dial plan for
> that.
>
>
> ;
> ;mailbox=1234 em default ; mailbox 1234 in voicemail context
> "default"
> disallow=all ; need to disallow=all before we can use
> allow=
> allow=ulaw ; Note: In user sections the order of codecs
>
>
> ; listed with allow= does NOT matter!
> allow=alaw
> allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
> allow=g729 ; Pass-thru only unless g729 license obtained
>
>
> ;callingpres=allowed_passed_screen ; Set caller ID presentation
> ; See doc/callingpres.txt for more
> information
>
> ;****
> ;**** Comunicação entre asterisks
> ;****
>
> [asterisk1]
>
>
> type=friend
> secret=welcome
> context=asterisk1_incoming
> host=dynamic
> disallow=all ; need to disallow=all before we can use
> allow=
> allow=ulaw ; Note: In user sections the order of codecs
>
>
> ; listed with allow= does NOT matter!
> allow=alaw
> allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
> allow=g729 ; Pass-thru only unless g729 license obtained
>
>
>
> *****************************************************************
> extensions.conf
>
> [phones]
> include=>internal
> include=>remote
>
>
> [internal]
> exten=>_1x,1,NoOp()
> exten=>_1x,n,Dial(SIP/${EXTEN},30)
>
>
> exten=>_1x,n+101,Hangup()
>
> [remote]
> ;exten=>_2x,1,NoOp()
> exten=>_2x,1,Dial(SIP/asterisk1/${EXTEN})
> exten=>_2x,n+101,Hangup()
>
> [asterisk1_incoming]
> exten=>_1x,1,DISA(no-password,internal)
>
>
> exten=>_3x,1,DISA(no-password,remote)
> exten=>_1x,102,Hangup()
> exten=>_3x,102,Hangup()
>
>
>
>
> _______________________________________________
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>
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>
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>
--
------------
Itamar Reis Peixoto
e-mail/msn: itamar em ispbrasil.com.br
sip: itamar em ispbrasil.com.br
skype: itamarjp
icq: 81053601
+55 11 4063 5033
+55 34 3221 8599
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