[Freeswitch-users] Freeswitch create bad SIP header

[Michael Collins]

What are you trying to accomplish? If you just want to send unauth'd calls
to/from each box that doesn't require a gateway in FreeSWITCH. However, if
you want the FreeSWITCH box to register to the Asterisk box then you'll
need a gateway.

If you just want to send a call from FreeSWITCH to Asterisk then you can
use a simple dialstring in your bridge line:

<action application="bridge" data="sofia/internal/1234 at asterisk.example.com
"/>

-MC


On Thu, May 16, 2013 at 7:44 AM, ik <idokan at gmail.com> wrote:

> Hello,
>
> I'm new to freeswitch, and I did google this, but can't find an answer
> that works for me.
>
> I have created the following configuration (both at Asterisk and FS sides):
> https://gist.github.com/ik5/1a0767f69b8065e1c8b8
>
> * The IP addresses are hidden and replaced with "asterisk.example.com"
> and "freeswitch.example.com"
>
> As you can see there with the SIP header, freeswitch create the "From" and
> "To" field with the same IP/Domain address, and there is no "contact" field
> to make it contact properly.
>
> So what am I missing or doing wrong ?
>
> Thanks,
>
> Ido
>
> _________________________________________________________________________
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>
> 
> 
>
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-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
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