Thanks for pointing me in the right direction. What fixed it was adding a '*' which I used to have before but with all these changes I was making I forgot to add it back in. So doing this:

<condition field="${sip_to_user}" expression="^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).*$" break="never">

works
Thanks for the help
\RR


On Tue, May 25, 2010 at 6:10 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you probably have to remove the $ at the end to allow for the ;params 


On Tue, May 25, 2010 at 4:43 PM, RR <ranjtech@gmail.com> wrote:
Hi Anthony,

this is what I see in the debug:

Dialplan: sofia/external/16469NNNNNN Regex (FAIL) [DIDtest] ${sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~ /^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$/ break=on-false

and then it moves on to another dialplan xml file.

please note that info app shows: 

variable_sip_to_user: [011390NNNNNNNNNN;phone-context=+39]

shouldn't this actually match??

Oh and yes, the copy of FS is pretty old but this is a production system which gets 24 / 7 traffic so the upgrade is being pushed and pushed :(

you think this is simply because it's an old build?

Thanks
RR


On Tue, May 25, 2010 at 4:49 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
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@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
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Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.




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@208.xx.xxx.xxx:5060]

Thanks
RR


On Tue, May 25, 2010 at 4:12 PM, Michael Collins <msc@freeswitch.org> wrote:


On Tue, May 25, 2010 at 12:48 PM, RR <ranjtech@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@67.1x.xxx.xxx:5060;user=phone>;tag=SDru6fc01-gK0c10a887.
(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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