[Freeswitch-users] Direct inward dialling

RR ranjtech at gmail.com
Tue May 25 14:43:37 PDT 2010


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
>> 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 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
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100525/a9e271d9/attachment.html 


More information about the FreeSWITCH-users mailing list