[Freeswitch-users] Still can't dial gateway from ZAP phone.
Jim
jim at k4gvo.com
Mon Aug 23 16:55:43 PDT 2010
Michael Collins gave me a suggestions a while back:
I looked in mod_openzap.c and I didn't see any references to channel
variables. However, you have context and dialplan options. I suggest that
you create a dialplan context just for your FXS port(s). Try this. Create
conf/dialplan/fxs-ports.xml:
<include>
<context name="fxs-ports">
<extension name="outbound">
<condition field="destination_number" expression="^(.*)$">
<action application="set"
data="toll_allow=local,domestic,international"/>
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
</context>
</include>
Then in your openzap.conf.xml change the context for the analog span(s) with
the FXS ports:
<param name="context" value="default"/>
Restart FS after making these changes and then give it a shot. You should
see the call from the analog phone going into context "fxs-ports" and then
get transferred over to the default context where it will act like your SIP
phones because we manually set the ${toll_allow} chan var.
-MC
Unfortunately that did not work. The "default_gateway" variable used by
this line:
Dialplan: OpenZAP/1:1/17705550068 Action
bridge(sofia/gateway/${default_gateway}/17705550068)
ended up looking like:
EXECUTE OpenZAP/1:1/17705550068 bridge(sofia/gateway//17705550068)
whereas a successful dialout looks like:
EXECUTE sofia/internal/1002 at 192.168.2.51
bridge(sofia/gateway/gw4.telasip.com/17705550068)
Somehow the information in the directory/default/default.xml file never
got included and I'm not sure how to fix it.
Thanks for any guidance.
Jim.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/813e3f55/attachment.html
More information about the FreeSWITCH-users
mailing list