[AsteriskBrasil] DISA e Interligacao entre asterisk (via SIP)
César Davi Avila do Nascimento
cesargxn em gmail.com
Quinta Julho 9 13:13:01 BRT 2009
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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