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()
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