[Freeswitch-users] FS and Asterisk connectivity
Arnaldo de Moraes Pereira
ap at arnaldopereira.com
Mon Apr 14 18:10:28 PDT 2008
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"/>
> sip_profiles/default/asterisk.xml :
> <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"/>
> The problem I get with this is no audio either way. I am on rev 8099.
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
More information about the FreeSWITCH-users