On 25 October 2016 at 12:33, Brian West <brian@freeswitch.org> wrote:
On Tue, Oct 25, 2016 at 2:29 PM, jungle Boogie <jungleboogie0@gmail.com> wrote:
On 25 October 2016 at 10:46, Brian West <brian@freeswitch.org> wrote:
I would like to get community input on what concepts caused you the most headache and any input/ideas that would help me design some tutorials / videos that would help new users coming to FreeSWITCH.
How about a dialplan explanation and example to: a) support calling sip URIs. (don't know if this supported from vanilla config)
Yes there is an example of this in the vanilla config:
<extension name="sip_uri">
<condition field="destination_number" expression="^sip:(.*)$">
<action application="bridge" data="sofia/${use_profile}/$1"/>
</condition>
</extension>
Okay, you're on the right track. Now explain in simple steps in your guide what to do with this.
b) support for input sip uri calling. 888@conference.freeswitch.org works very well. How can people do the same?
The one in the vanilla config would allow you to dial anything sip:url and pass it to a bridge line, mostly used with port audio.
But how does it handle authenticaled/unauthenticated callers? Do acl rules need to be made? I'd really like to have jungle@example.com ring an end point/IVR without messy ACL rules. Is that possible?