se quiser que eu te explique uma maneira facil de fazer isto é só me ligar no sip: ita...@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 <cesar...@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=1...@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=1...@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=1...@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=1...@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() > > > > > _______________________________________________ > Openmoko Freerunner, primeiro telefone open source, disponível no Brasil > rodando o Android da Google. > http://www.neodroid.com > > Compre uma camiseta da AsteriskBrasil.org! > http://www.voipmania.com.br > > Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na > rede Freenode.net: #asterisk-br > _______________________________________________ > Lista de discussões AsteriskBrasil.org > AsteriskBrasil@listas.asteriskbrasil.org > http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil > -- ------------ Itamar Reis Peixoto e-mail/msn: ita...@ispbrasil.com.br sip: ita...@ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 _______________________________________________ Openmoko Freerunner, primeiro telefone open source, disponível no Brasil rodando o Android da Google. http://www.neodroid.com Compre uma camiseta da AsteriskBrasil.org! http://www.voipmania.com.br Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na rede Freenode.net: #asterisk-br _______________________________________________ Lista de discussões AsteriskBrasil.org AsteriskBrasil@listas.asteriskbrasil.org http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil