[Freeswitch-users] Direct inward dialling
Anthony Minessale
anthony.minessale at gmail.com
Tue May 25 15:10:54 PDT 2010
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
>>> 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
>>
>>
>
> _______________________________________________
> 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100525/c3488d49/attachment.html
More information about the FreeSWITCH-users
mailing list