But to have a transparent integration with VoIP and legacy, I cant make
users dial twice... or having to whait for Asterisks dialtone, and dial
the number.
I whant to dial the 74XXX from a PBX extension (74118 for example) and
the IP phone rings.
Asterisk just need to forward the 74XXX calls, thats why I think the
solution is close to this:
exten => _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
... but it always answers:
pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/1' sent into invalid
extension 's' in context 'default', but no invalid handler
Why is CAPI sending it to 's' if I explicitly write
"Dial(SIP/[EMAIL PROTECTED],30,r)" ??
João
Matt Riddell wrote:
Joao Pereira wrote:
Hello list,
Im writing my dial plan, in witch every SIP phone begins with 74 and
has more 3 numbers (like 74XXX).
So, I want to route all 74XXX calls to my sip channel. For this I
wrote this line:
exten => s,1,Dial(SIP/[EMAIL PROTECTED],30,r)
What is happening is that capi is sending it to s.
You will need to either set up an IVR, asking which number to send it to.
So, you would do the following:
exten => s,1,Answer()
exten => s,2,Background(pls-entr-extn)
exten => _74XXX,1,Dial(SIP/${EXTEN})
exten => _74XXX,2,Goto(s|1)
exten => _74XXX,102,Goto(s|1)
You will obviously need to record the pls-entr-extn sound.
You can do this by making an exten like this:
exten => 678,1,Record(pls-entr-extn)
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