[Freeswitch-users] What concepts in FreeSWITCH did you struggle with when you first started?

[jungle Boogie]

On 25 October 2016 at 12:33, Brian West <brian at freeswitch.org> wrote:
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>
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> On Tue, Oct 25, 2016 at 2:29 PM, jungle Boogie <jungleboogie0 at gmail.com> wrote:
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>> On 25 October 2016 at 10:46, Brian West <brian at freeswitch.org> wrote:
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>> > 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.
>> >
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>> How about a dialplan explanation and example to:
>> a) support calling sip URIs. (don't know if this supported from vanilla config)
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> Yes there is an example of this in the vanilla config:
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>     <extension name="sip_uri">
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>       <condition field="destination_number" expression="^sip:(.*)$">
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>         <action application="bridge" data="sofia/${use_profile}/$1"/>
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>       </condition>
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>     </extension>
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>

Okay, you're on the right track. Now explain in simple steps in your
guide what to do with this.

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>> b) support for input sip uri calling. 888 at conference.freeswitch.org
>> works very well. How can people do the same?
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> 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.
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But how does it handle authenticaled/unauthenticated callers? Do acl
rules need to be made? I'd really like to have jungle at example.com ring
an end point/IVR without messy ACL rules. Is that possible?