[OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?

Jesus Rodriguez jesusr at voztele.com
Wed Apr 2 18:21:59 CEST 2008


Hola Iñaki,

El 02/04/2008, a las 18:01, Iñaki Baz Castillo escribió:
> El Wednesday 02 April 2008 15:48:10 Jesus Rodriguez escribió:
>> Hola Iñaki,
>>
>> El 02/04/2008, a las 17:40, Iñaki Baz Castillo escribió:
>>> El Wednesday 02 April 2008 15:35:40 Jesus Rodriguez escribió:
>>>>> The info about the called PSTN number is just available in "To"
>>>>> header, so a
>>>>> way to get different behaviour for each associated PSTN number is
>>>>> matching "To" URI.
>>>>> Is common to do it? which other alternatives are there?
>>>>
>>>> You can also add multiple aliases to the same user. All INVITEs to
>>>> different aliases will be sent to the resolved user.
>>>
>>> Yes, but after the "lookup" the RURI username will be definitively  
>>> the
>>> username the client sent in the REGISTER's "Contact", so the INVITE
>>> will not
>>> change, will it?
>>
>> The RURI will be changed by the registered Contact value. The To:
>> header is not modified.
>
> Exactly, but a client will have (probably) just one sip account  
> registered, so
> just one entry will appear in "location" table for him AoR.
>
> Imagine a user "clientX at sip_proxy" using an Asterisk and registering  
> with:
>  REGISTER sip:sip_proxy SIP/2.0
>  To: <sip:clientX at sip_proxy>
>  Contact: <sip:s at ip_asterisk>
>
> It will appear in location table with "Contact=sip:clientX at IP".
>
> Suppose clientX has two PSTN numbers associated in a ENUM entry:
>  +34999000111
>  +34999000222
>
> When it receives a call from PSTN to +34999000222 the INVITE  
> arriving to
> Asterisk will be:
>
>  INVITE sip:s at ip_asterisk SIP/2.0
>  To: <sip:+34999000222 at sip_proxy>
>
> So the info about the real PSTN  number dialed by the call  
> originator just
> remains in the "To" header. So the client (Asterisk) must read the  
> "To"
> header in order to have a behaviour different for +34999000111 and
> +34999000222 (or adding in OpenSer other custom header containing  
> also dialed
> number 999000222 as Juha suggests).
> I can't understand how using aliases in OpenSer can help here. ¿?


Yes, in theses cases you have to relay on To: header value. The  
problem is that not all "multiport" devices are "multi account" as  
Asterisk is and you need to use aliases to send all different DIDs to  
the same SIP account... some of these "multiport" devices can read the  
To: value and send the call to the right place even registering only  
one account.


Saludos
JesusR.

------------------------------------
Jesus Rodriguez
VozTelecom Sistemas, S.L.
jesusr at voztele.com
http://www.voztele.com
Tel. 902360305
-------------------------------------








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