[asterisk-dev] Fwd: Inquiry:Asterisk sip server?

[hadi motamedi]

Thank you very much for your reply . I asked on the users mailing list and
waited for a long time but didn't receive any response . So I thought if it
must be asked from the developers mailing list . Sorry for my mistake .
Thank you in advance



On Tue, Dec 15, 2009 at 8:07 PM, John Todd <jtodd at digium.com> wrote:

>
> On Dec 15, 2009, at 4:02 AM, hadi motamedi wrote:
>
> > ---------- Forwarded message ----------
> > From: hadi motamedi <motamedi24 at gmail.com>
> > Date: Sat, Dec 12, 2009 at 9:44 AM
> > Subject: Inquiry:Asterisk sip server?
> > To: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users at lists.digium.com
> > >
> >
> >
> > Dear All
> > I have an application that calls for Asterisk sip configuration to
> > be able to communicate with external sip server . My Asterisk 3.1.14
> > has been installed on Debian 3.1 server and the external sip server
> > is @192.168.0.10 , the same subnet as my Debian server
> > @192.168.0.2  . At now , the configuration is in such a way that the
> > call attempts reaching to my Asterisk are being routed internally ,
> > based on my Asterisk extensions.conf settings . I need to change the
> > current configuration in such a way that the voip call attempts to
> > be routed toward the external sip server at 192.168.0.10 for the call
> > routing purposes . Can you please help me how I am expected to
> > modify my Asterisk configuration to do the job ?
> > Regards
> > H.Motamedi
>
>
> Hadi -
>   I would suggest asking this question on the "asterisk-users"
> mailing list, as the asterisk-dev list is for discussion only of
> internal programming, code, bug or other development-related issues.
> You will find that the asterisk-users list also has a much larger
> number of active participants who may have suggestions on your
> problem.  Thanks!
>
> (As a side note, you may wish to create a small web page which
> includes your configuration files and perhaps a drawing of what you
> wish to accomplish, as I suspect the description you have given is not
> sufficient for someone to fully answer your question.  Include the
> link to that web page description in your post to asterisk-users.)
>
> JT
>
> ---
> John Todd                       email:jtodd at digium.com<email%3Ajtodd at digium.com>
> Digium, Inc. | Asterisk Open Source Community Director
> 445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
> direct: +1-256-428-6083         http://www.digium.com/
>
>
>
>
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