[OpenSIPS-Users] Delete alias references in SIP messages

[Guillaume Lacroix]

Hi Bogdan,

Thanks for your time. I think I will follow the second option and just 
wait for a while till opensips get the feature. In the meantime, I'll 
use some workaround (use of black numbers or use a B2BUA).

Regards,
Guillaume

Bogdan-Andrei Iancu a écrit :
> 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
>>>>
>>>>   
>>>
>>>
>>
> 
>