[Freeswitch-users] Direct inward dialling
RR
ranjtech at gmail.com
Tue May 25 15:34:04 PDT 2010
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 at 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 at 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 at 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 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
>>>> tel: 0811 46 26 26
>>>> www.ipeva.fr - www.ipeva-studio.com
>>>>
>>>> *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
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>>>> 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 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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>>>>>
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>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
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>>>
>>>
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>>
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>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
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>
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