There might also be a way to configure the SIP phone itself to use a
more correct format for the invite, thus eliminating the need to adjust
your dialplan.
--James
On 06/27/2013 12:30 PM, Shamus Rask wrote:
This isn't AstLinux specific, however I can no longer find any
helpful, active Aster
Head over to the asterisk irc chan on free node.
On Thursday, June 27, 2013, David Kerr wrote:
> There are of course a bunch of syntax errors in the example I used !!!
> I'm sure you can figure it out.
>
> David
>
>
> On Thu, Jun 27, 2013 at 1:27 PM, David Kerr 'cvml', 'da...@kerr.net');>
> > w
There are of course a bunch of syntax errors in the example I used !!! I'm
sure you can figure it out.
David
On Thu, Jun 27, 2013 at 1:27 PM, David Kerr wrote:
> You could use the asterisk FILTER() function to remove everything except
> numbers 0-9 then either goto(FILTER(0-9,${exten})) or
>
You could use the asterisk FILTER() function to remove everything except
numbers 0-9 then either goto(FILTER(0-9,${exten})) or
dial(${trunk}/$FILTER(0-9,${exten}),bla,bla)
David
On Thu, Jun 27, 2013 at 12:30 PM, Shamus Rask wrote:
> This isn't AstLinux specific, however I can no longer find an
This isn't AstLinux specific, however I can no longer find any helpful, active Asterisk forums. The digium ones just seem dead…I'm currently running the latest version of AstLinux with Asterisk v1.8.21. I have an existing dialplan that allows for basic NANP dialing in the form of: exten => _1NXXNXX
Hi Lonnie,
Thx for the info. Looks like a great solution.
lach
On Wed, Jun 26, 2013 at 3:05 PM, Lonnie Abelbeck
wrote:
> Lach,
>
> That box has plenty of power for that application, even if you did
> transcoding on the calls.
>
> As a quick test I ran 4 concurrent calls (ulaw no transcoding
