[asterisk-users] Call Forwarding in SIP.conf

[Tim St. Pierre]

Check your Dial() string to make sure that you haven't mistyped and put 
gafachi-o instead of gafachi-out.  Specifiying the full host name will also 
work.

As a hint, you can refresh these changes with out restarting your server (and 
therefore without disrupting any calls in progress)

extensions reload will refresh the extensions file
reload will reload all your configs
sip reload will reload only sip configs (and re-register everything)

Very handy when working on an active machine.


On September 8, 2006 14:19, broadbandvoice at comcast.net wrote:
> Tim, this is the way I have Gafachi set up in sip.conf and works well with
> channels that have an ATA attached to it but not the virtual one. I have
> changed the host in extensions.conf to the xxxxxxxxxxxx.sip.gafachi.com.
> But I have calls on the server and cannot restart it yet. I'll keep you
> posted and thanks for the feedback.
>
> [gafachi-out]
> type=peer
> secret=xxxxxxxxxx
> username=xxxxxxxxx
> fromuser=xxxxxxxxxxxxx
> fromdomain=xxxxxxxxxxxxxxx
> host=xxxxxxxxxxxx.sip.gafachi.com
> ;usereqphone=yes                ; This provider requires ";user=phone" on
> URI ;nat=yes
> rtptimeout=60
> dtmfmode=rfc2833
>
>
> -------------- Original message --------------
> From: "Tim St. Pierre" <tim at communicatefreely.net>
>
> > _______________________________________________
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-- 
Tim St. Pierre

IP telephony specialist
sip://5101@communicatefreely.net
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
tim at communicatefreely.net
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