[Freeswitch-users] can't pass full sip url to dialplan
Henry Huang
red.rain.seven at gmail.com
Sat Aug 22 10:35:01 PDT 2009
Michael:
Thank you for making it in "for dummies" format. :P
These are really nice tips I can use. thanks.
On Sat, Aug 22, 2009 at 11:35 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>
> 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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>>
>
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>
--
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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