On Mon, Apr 14, 2008 at 4:49 PM, Brian Snipes <[EMAIL PROTECTED]> 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 > > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
