On Mon, Apr 14, 2008 at 4:49 PM, Brian Snipes <bsnipes@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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