[OpenSIPS-Users] Delete alias references in SIP messages

[Bogdan-Andrei Iancu]

Hi Guillaume,

I see the problem - of course, except the case when you want to use mid 
media serve (Asterisk, Yate, freeswitch) as a B2BUA, right now you 
cannot change the TO header without breaking the dialog. But, there are 
two solutions:
    1) upload a feature request on the tracker for implementing in 
opensips TO changing (in the same manner as for FROM)
    2) we plan to implement a signalling B2BUA in opensips and probably 
this will solve your problem, but it will take some time.

Regards,
Bogdan

Guillaume Lacroix wrote:
> Hi Bogdan,
>
> Thanks for the hints. Here is the stuff : we would like to give 
> premimum numbers (08XX...) to some of our customers.
>
> These premium numbers are provided by an external carrier A. This 
> carrier translate the number to one geographical number we choose. 
> This geographical number will be an alias of a SIP account.
>
> So : Regular phone in the PSTN -> 08XX (carrier A) -> 01XX (us - 
> customer 1) -> SIP account (us - customer 1)
>
> The problem is that the carrier A doesn't give any information that 
> the regular phone has dialed the 08XX number (it is just like the 
> regular phone dialed the 01XX directly). So, just by tracing the SIP 
> messages, the customer 1 could easy know the translated number.
>
> Since we can give some payback on incoming trafic for these numbers, 
> we would like to avoid that (so, except carrier 1 and us, no one has 
> to know this translated 01XX number). So the best would be to be able 
> to modify any reference in SIP messages of the 01XX number to 08XX 
> number (so that we can keep a trace that the number dialed should have 
> been 08XX).
>
> Thanks and regards,
> Guillaume
>
> Bogdan-Andrei Iancu a écrit :
>> Hi Guillaume,
>>
>> Changing TO header is more or less against the RFC3261 and you should 
>> not do it. More or less the same discussion as for the FROM header.
>>
>> Even if in openSIPS you can change the FROM without violating the RFC 
>> (doing and undoing the change for all the sequential request inside 
>> the dialog), for TO there is no such mechanism.
>>
>> Why is so critical to remove it from the alias from the TO header ? 
>> Understanding the cause may help me to give you some ideas ;)
>>
>> Regards,
>> Bogdan
>>
>> Guillaume Lacroix wrote:
>>> Hello,
>>>
>>> I would like to know if there an easy way to remove any reference to 
>>> the SIP alias in the SIP messages for some incoming calls and to 
>>> replace them with the SIP username.
>>>
>>> Roughly, a user dial the number 012345 from the PSTN. The call goes 
>>> to a PSTN GW and is then sent to OpenSER through an INVITE message 
>>> (INVITE 012345 at domain...). The To header also contains 
>>> 012345 at domain. After the look up in the alias DB, the alias is 
>>> replace by the SIP username, but it remains in the To header field.
>>>
>>> What I would like to do is to remove completely the alias reference 
>>> and replace it with the username, not only in the INVITE but also in 
>>> any following messages (so that the messages received by the UA 
>>> never contains any reference to the alias) I try to use 
>>> append_hf/remove_hf on the To header, it works, but the alias still 
>>> appears in some other messages (ACK..).
>>>
>>> Does anyone know if there is a way to achieve that (not for all 
>>> incoming number but only the number starting with 0123 for example) ?
>>>
>>> Thanks and regards,
>>> Guillaume
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>   
>>
>>
>