Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=g729
[1000]
type=friend
username=1000
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
[2000]
type=friend
username=2000
secret=password2
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
extension.conf
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/1000
PHONE2=SIP/2000
[international]
ignorepat => 88
exten=> _1N1NXXNXXXXXX,1,Dial ???????
[internal]
include => local-sip
[local-sip]
exten => 1000,1,Dial(${PHONE1},40,t)
exten => 1000,2,Hangup
exten => 2000,1,Dial(${PHONE2},40,t)
exten => 2000,2,Hangup
exten => 1001,1,Dial(${PHONE3},40,t)
exten => 1001,2,Hangup
i want user to dial 88 and they will get a tone and dial US or UK
number from local-sip context.
the provider only gave me a IP to route my SIP traffic and needs no
registration, can any please help me how to write the International
context in extension.conf also what i shld do in sip.conf..
my audiocodec supports g729 and g723 codec so do i need to aqquire
license for G729 from digium, if yes then why?
last if possible can you also please tell me wht i need to add on my
context so user can while in calling put the call on hold and transfer
to another sip phone.
I thankyou all for reading this mail and i hope someone will be kind
enough to help.
Best regards,
Imran
Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=g729
[1000]
type=friend
username=1000
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
[2000]
type=friend
username=2000
secret=password2
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
extension.conf
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/1000
PHONE2=SIP/2000
[international]
ignorepat => 88
exten=> _1N1NXXNXXXXXX,1,Dial ???????
[internal]
include => local-sip
[local-sip]
exten => 1000,1,Dial(${PHONE1},40,t)
exten => 1000,2,Hangup
exten => 2000,1,Dial(${PHONE2},40,t)
exten => 2000,2,Hangup
exten => 1001,1,Dial(${PHONE3},40,t)
exten => 1001,2,Hangup
i want user to dial 88 and they will get a tone and dial US or UK
number from local-sip context.
the provider only gave me a IP to route my SIP traffic and needs no
registration, can any please help me how to write the International
context in extension.conf also what i shld do in sip.conf..
my audiocodec supports g729 and g723 codec so do i need to aqquire
license for G729 from digium, if yes then why?
last if possible can you also please tell me wht i need to add on my
context so user can while in calling put the call on hold and transfer
to another sip phone.
I thankyou all for reading this mail and i hope someone will be kind
enough to help.
Best regards,
Imran
Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=g729
[1000]
type=friend
username=1000
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
[2000]
type=friend
username=2000
secret=password2
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
extension.conf
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/1000
PHONE2=SIP/2000
[international]
ignorepat => 88
exten=> _1N1NXXNXXXXXX,1,Dial ???????
[internal]
include => local-sip
[local-sip]
exten => 1000,1,Dial(${PHONE1},40,t)
exten => 1000,2,Hangup
exten => 2000,1,Dial(${PHONE2},40,t)
exten => 2000,2,Hangup
exten => 1001,1,Dial(${PHONE3},40,t)
exten => 1001,2,Hangup
i want user to dial 88 and they will get a tone and dial US or UK
number from local-sip context.
the provider only gave me a IP to route my SIP traffic and needs no
registration, can any please help me how to write the International
context in extension.conf also what i shld do in sip.conf..
my audiocodec supports g729 and g723 codec so do i need to aqquire
license for G729 from digium, if yes then why?
last if possible can you also please tell me wht i need to add on my
context so user can while in calling put the call on hold and transfer
to another sip phone.
I thankyou all for reading this mail and i hope someone will be kind
enough to help.
Best regards,
Imran
Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=g729
[1000]
type=friend
username=1000
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
[2000]
type=friend
username=2000
secret=password2
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
extension.conf
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/1000
PHONE2=SIP/2000
[international]
ignorepat => 88
exten=> _1N1NXXNXXXXXX,1,Dial ???????
[internal]
include => local-sip
[local-sip]
exten => 1000,1,Dial(${PHONE1},40,t)
exten => 1000,2,Hangup
exten => 2000,1,Dial(${PHONE2},40,t)
exten => 2000,2,Hangup
exten => 1001,1,Dial(${PHONE3},40,t)
exten => 1001,2,Hangup
i want user to dial 88 and they will get a tone and dial US or UK
number from local-sip context.
the provider only gave me a IP to route my SIP traffic and needs no
registration, can any please help me how to write the International
context in extension.conf also what i shld do in sip.conf..
my audiocodec supports g729 and g723 codec so do i need to aqquire
license for G729 from digium, if yes then why?
last if possible can you also please tell me wht i need to add on my
context so user can while in calling put the call on hold and transfer
to another sip phone.
I thankyou all for reading this mail and i hope someone will be kind
enough to help.
Best regards,
Imran