[Freeswitch-users] FS and Asterisk connectivity

[Arnaldo de Moraes Pereira]

On Mon, Apr 14, 2008 at 4:49 PM, Brian Snipes <bsnipes at snipes.org> wrote:
> I wish to connect FS to * for interoperability testing and can't seem to
>  get my configs correct.  Has anyone done this already and if so can you
>  post your configs?

My configs are like yours, except for three things:
  1. my asterisk is configured on outbound profile, instead of default's
  2. FS registers to my asterisk
  3. My rev: 8081

Are you sure the sampling rate are ok for both legs ? Besides that and
comfortable noise generation (turned off, as: <param
name="supress-cng" value="true"/>), I can't think of anything else.

My 8081 rev is working nicely with asterisk, maybe I should update and
see what happens.

>
>  I can call from FS to the asterisk side via a dialplan entry where
>  ^9(.*)$ passes to a gateway I've setup:
>
>     <extension name="out9_asterisk">
>       <condition field="destination_number" expression="^9(.*)$">
>         <action application="bridge" data="sofia/gateway/asterisk/$1"/>
>       </condition>
>     </extension>
>
>  sip_profiles/default/asterisk.xml :
>
>  <include>
>   <gateway name="asterisk">
>   <param name="username" value="freeswitch"/>
>   <param name="realm" value="x.x.x.x"/>
>   <param name="password" value="password"/>
>   <param name="register" value="false"/>
>   </gateway>
>  </include>
>
>  The problem I get with this is no audio either way.  I am on rev 8099.
>
>  TIA,
>  Brian
>
>
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