[asterisk-users] Call Forwarding in SIP.conf

[William Piper]

whatever the did is needs to be put in the extensions.conf & told to dial
your cellphone.
Example:

exten => _011123445566,1,Dial,SIP/1234567890 at 1.2.3.4

assuming that your using a SIP carrier, replace 1234567890 with your
cellphone & 1.2.3.4 with the carrier's IP or carriers context name in
sip.conf.

bp

On 9/8/06, broadbandvoice at comcast.net <broadbandvoice at comcast.net> wrote:
>
>  I'm using it for virtual numbers. I have international virtual number
> from a DID provider and want to forward it to my cell phone.
>
> In Sip.conf I have the channel
>
>
> [4305]
>
> type=friend
>
> user=4305
>
> secret=xxxxxxxx
>
> ;context=from-sip
>
> callerid=
>
> host=dynamic
>
> nat=yes
>
> canreinvite=no
>
> dtmfmode=rfc2833
>
> ;incominglimit=1
>
> ;mailbox=1234 at default
>
> ;disallow=all
>
> ;allow=ulaw
>
> ;allow=alaw
>
> ;allow=g723.1                   ; Asterisk only supports g723.1 pass-thru!
>
>
>
> and in extensions.conf I have
>
>
>
> exten => 4305,1,Dial(SIP/4305,120,rt)      ; permit transfer
>
>
>
> This had worked in the past when I forwarded it through the Linksys ATA
> but now have run out of ATA's.
>
>
>
>
> -------------- Original message --------------
> From: "Tim St. Pierre" <tim at communicatefreely.net>
>
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