[Freeswitch-users] Direct inward dialling

[David Ponzone]

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
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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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