[Freeswitch-users] can't pass full sip url to dialplan

[Michael Collins]

On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang <red.rain.seven at gmail.com>wrote:

> Brian:
>
> Oh, and again, if it's not passing it to the dialplan. I had suggested to
> remove the sample "sip uri" extension in the default.xml dialplan. because
> no one can reach the dialplan with prefix "sip:" because sofia is going to
> remove that prefix.


Well, this isn't entirely accurate. Like Mike J said, if you dialed
something like this at the CLI:

pa call sip:user at domain.com <sip%3Auser at domain.com>

Then you'd need the dialplan entry that handles the SIP URI.

Going back to the original question...
X-Lite dials 1009 at 4.2.2.2 correct?
But you're saying that the dialplan simply sees "1009" as the destination
number? I'm looking at the pastebin (10089) and trying to figure out exactly
what is happening. All I can see is that you have a context named "Global"
so I'm assuming you've made at least some modifications to the default
dialplan. Can you pastebin that whole context?

The other thing that you should probably do is create an extension in this
global context that routes a call to the info application. You could do
something like this so that "9992" would do an info dump:
<extension name="info">
  <condition field="destination_number" expression="^(9992)$">
    <action application="info"/>
  </condition>
</extension>

Then reloadxml and make a call to 9992 from your X-Lite client. The CLI will
have a dump and you'll see all sorts of variables listed. Many of those are
available for you to use for condition matches and routing in the dialplan.

Let us know how the info application does in giving you information about
the A leg of the call.
-MC

>
>
> <!-- dial via SIP uri -->
>     <extension name="sip_uri">
>       <condition field="destination_number" expression="^sip:(.*)$">
>         <action application="bridge" data="sofia/${use_profile}/$1"/>
>       </condition>
>     </extension>
>
> On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Because the dial plan is technology agnostic... you have been told
>> more than once it won't pass it to the dialplan from mod_sofia...
>>
>> /b
>>
>> On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
>>
>> > Brian:
>> >
>> > but why can't I pass "sip:" to dialplan? seems like it's being
>> > truncated by sofia..
>> > Can you confirm that?
>>
>>
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>
>
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>
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