Consider using a sip proxy server such as OpenSIPS or Kamailio.
Regards,
Ali Pey
On Mon, Dec 10, 2012 at 12:59 PM, John Gilbert
wrote:
> I have a non-standard SIP client that I am trying to integrate with an
> Asterisk 10 server.
>
> This client requires that it register with
SIP peer doesn't need to be registered. It uses the host value to talk to
the peer.
Regards,
Ali Pey
On Thu, Nov 15, 2012 at 1:15 PM, Face wrote:
> On Wed, Nov 14, 2012 at 7:42 PM, Steve Edwards
> wrote:
> >> On Wed, 14 Nov 2012, Face wrote:
> >>
> >>
You can also consider using a proxy server such as opensips or Kamailio.
They would enable you to do much more at the signalling level and many
other advantages such as better security or nat traversal.
Regards,
Ali Pey
On Wed, Nov 14, 2012 at 11:42 AM, Steve Edwards
wrote:
> On Wed, 14
We have also used enswitch from Integrics and I totally recommend it. It
does all the things you ask for and some more.
Regards,
Ali Pey
On Tue, Nov 13, 2012 at 9:48 PM, Alistair Cunningham <
acunning...@integrics.com> wrote:
> Hello Marshall,
>
> Please see Ensw
I have also uploaded my presentation here:
http://www.slideshare.net/alipey/astricon-2012-redundancy-and-high-availability?from=share_email
It's on Redundancy and high availability using OpenSIPS/Kamailio.
Regards,
Ali Pey
On Tue, Nov 13, 2012 at 4:24 AM, Lenz Emilitri wrote:
>
d ulaw with no luck. I read the SIP_CODEC value and it
is set properly.
Any suggestions/ideas?
Thanks,
Ali Pey
On Thu, Nov 1, 2012 at 12:02 PM, qasimak...@gmail.com
wrote:
> exten => _X.,1,NoOP(G711 CoDec)
> exten => _X.,n,Set(SIP_CODEC=g711)
> exten => _X.,n,Dial(...)
>
>
call was routed to T1 card over
Dahdi but G729 if the call was going to another sip client.
Thanks,
Ali Pey
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introdu
