[Freeswitch-users] Direct inward dialling

Tue May 25 13:49:45 PDT 2010

did you turn up the debug (press f8 or type console loglevel debug)
The debug logs will show the data being passed into the regex and the
results.

P.S.
I hope only your example is from years ago and not your copy of FS.

On Tue, May 25, 2010 at 3:40 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> Which means there is no @ in the sip: part of the SIP To field. Only in the
> phone-context part.
> FS uses the @ to split the strings into pieces, and then in your case, it
> fails as one is missing.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
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>
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>
> Le 25/05/2010 à 22:28, RR a écrit :
>
> Hi Guys,
>
> Thanks for the quick feedback
>
> David, no we're getting the full URI with the domain part intact, just
> nothing before the "<" braces
>
> Michael, I already tried the info app and we get
>
> variable_sip_to_user: [011390NNNNNNNNNN;phone-context=+39]
> variable_sip_to_uri: [011390NNNNNNNNNN;phone-context=+39 at 208.xx.xxx.xxx
> :5060]
>
> Thanks
> RR
>
>
> On Tue, May 25, 2010 at 4:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Tue, May 25, 2010 at 12:48 PM, RR <ranjtech at gmail.com> wrote:
>>
>>> Hello I want to follow up on this example from YEARS ago. I had tried
>>> using the variable "destination_number" but that didn't work, and I figured
>>> that it was because the To: header doesn't have the destination_number but
>>> has just the URI, so I thought I'd use sip_to_user instead.
>>>
>>> We have calls coming in with the following info in the INVITE
>>>
>>> From: "16469NNNNNN" <
>>> sip:16469NNNNNN;phone-context=+1 at 67.1x.xxx.xxx:5060;user=phone
>>> >;tag=SDru6fc01-gK0c10a887.
>>> To: <
>>> sip:011390NNNNNNNNNN;phone-context=+39 at xxx.xxx.xxx.xxx:5060;user=phone>.
>>> (N and x are obviously being masked for privacy)
>>>
>>> I use this info in the dialplan like so
>>>
>>> <include>
>>>   <extension name="DIDtest">
>>>     <condition field="ani" expression="^(\+?|\+1?|1?)(6469NNNNNN).*$"
>>> break="never">
>>>         <action application="set" data="effective_caller_id_number=$2"/>
>>>         <action application="set" data="effective_caller_id_name=$2"/>
>>>     </condition>
>>>     <condition field="${sip_to_user}"
>>> expression="^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$" break="never">
>>>         <action application="set" data="continue_on_fail=false"/>
>>>         <action application="set" data="hangup_after_bridge=true"/>
>>>         <action application="set" data="domain_name=$${domain}"/>
>>>         <action application="set" data="bypass_media=true"/>
>>>         <action application="bridge"
>>> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$2"/>
>>>     </condition>
>>>   </extension>
>>> </include>
>>>
>>> However, the calls aren't passing the condition in this dialplan and thus
>>> not being forwarded to "blade2" server. In fact, even the 011 is not being
>>> stripped off.
>>>
>>> What am I doing wrong?
>>>
>>
>> Create a quick test extension that only does an info dump. (See 9992 in
>> default.xml for an example.) Make a call, look at the info dump, and make
>> sure that what you think you are getting is really what you are getting. :)
>>
>> -MC
>>
>>
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