[OpenSIPS-Users] SIP trunk provider lab

[Bogdan-Andrei Iancu]

Hi Robert,

Robert Borz wrote:
> Hi Bogdan,
>
> I tried it and it works great! Thanks again for your help. :-)
>   
glad it worked.
> But one thing... how can I achieve this the other way around?
> userXY at domain.de should be able to place calls, which should look as originated from one of its "alias" accounts.
>
> I think it must be as easy as your workaround... but I'm currently a little lost. :-/ Just not my day...
>   
First you need somehow to determine what alias you want to use as new 
identity. After that, use uac_replace_from() to change the From hdr (if 
you have SIP destinations) or try using PAI/RPID hdr if the destination 
is a GW.

Regards,
Bogdan
>
> Regards,
> Robert
>
> -----Original Message-----
> From: bogdan at voice-system.ro [mailto:bogdan at voice-system.ro] 
> Sent: Tuesday, April 21, 2009 11:51 AM
> To: robert.borz at web.de
> Cc: home at elnour.biz; users at lists.opensips.org
> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>
> What you can do is:
>
> 1) before applying aliases, save the RURI into an AVP
> 2) do alias ->userXY, do whatever other routing stuff, including 
> lookup("location");
> 3) just before sending out the request, do: if $du (destination URI) is 
> empty, copy the current RURI into $du ($du = $ru); (id $du already set, 
> skip that step). After that, copy the stored AVP into RURI and send it out.
>
> more or less you will the desitnation URI to point to userXY and put in 
> RURI the alias stuff..
>
> Regards
> Bogdan
>
> Robert Borz wrote:
>   
>> Hi Bogdan,
>>
>> yes, that's exactly what I want... :-)
>>
>>
>> Regards,
>> Robert.
>>
>> -----Original Message-----
>> From: bogdan at voice-system.ro [mailto:bogdan at voice-system.ro] 
>> Sent: Tuesday, April 21, 2009 11:38 AM
>> To: robert.borz at web.de
>> Cc: home at elnour.biz; users at lists.opensips.org
>> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>>
>> Hi Robert,
>>
>> you mean, at the end, you want to sent it to userXY at domain.de phone, but 
>> with 1000 at domain.de in RURI, right ?
>>
>> Regards,
>> Bogdan
>>
>> Robert Borz wrote:
>>   
>>     
>>> Hi Bogdan,
>>>
>>> could you be more specific here, please?
>>>
>>> I want to setup a similar configuration. I want to forward let's say 300 numbers (sip accounts) to a single SIP account without loosing the R-URI, because the box receiving the forwarded calls should be able to distinct which number was dialed. What do I have to insert in the alias table to do this:
>>>
>>> 1000 at domain.de
>>> 1001 at domain.de
>>> 1002 at domain.de	-- forward to ---> userXY at domain.de
>>> ...
>>> 1010 at domain.de
>>>
>>> Another thing I'm currently completely lost in is how to handle the outgoing part. userXY at domain.de should be able to place outgoing calls with the originator set to one of the 1000 at domain.de ... 1010 at domain.de addresses.
>>>
>>> How can I achieve this behaviour? Thanks a lot!
>>>
>>>
>>> Regards,
>>> Robert
>>>
>>>
>>> -----Original Message-----
>>> From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
>>> Sent: Monday, March 16, 2009 9:14 AM
>>> To: home at elnour.biz
>>> Cc: users at lists.opensips.org
>>> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>>>
>>> Hi Elnor,
>>>
>>> home at elnour.biz wrote:
>>>   
>>>     
>>>       
>>>> Hi,
>>>>
>>>> thank you for your response!
>>>> correct me if I'm wrong, but aren't we going to loose dest phone number if 
>>>> we use dbaliases?
>>>>   
>>>>     
>>>>       
>>>>         
>>> Not necessary - depends of how you define the alias. For ex, you can do:
>>>     DID at server -> DID@ trunk
>>>
>>> preserve the username part and change only the domain to point to the trunk.
>>>
>>>
>>> Also, if you have blocks of DIDs which are easy to detect based on 
>>> regexp, you may consider using the dialplan module:
>>>        http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html
>>>
>>> See example 1.4.1.2 - 
>>> http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html#id227187
>>>
>>> Regards,
>>> Bogdan
>>>
>>>   
>>>     
>>>       
>>>> Also please provide hints on ENUM, how do I use it for this purpose?
>>>>
>>>> Thank you.
>>>>
>>>> Elnour
>>>>
>>>>
>>>> ----- Original Message ----- 
>>>> From: "Iñaki Baz Castillo" <ibc at aliax.net>
>>>> To: <users at lists.opensips.org>
>>>> Sent: Saturday, March 14, 2009 3:59 PM
>>>> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>>>>
>>>>
>>>>   
>>>>     
>>>>       
>>>>         
>>>>> El Sábado, 14 de Marzo de 2009, home at elnour.biz escribió:
>>>>>     
>>>>>       
>>>>>         
>>>>>           
>>>>>> if we have a SIP subscriber registered with a user name companyA which 
>>>>>> has
>>>>>> 8 DID phone lines from PSTN say 5551001-08, our setup should dynamicaly
>>>>>> route all incomming calls to the "location" of the regisration.
>>>>>>
>>>>>> Is this possbile? I mean is this possible to achive dynamicaly?
>>>>>>       
>>>>>>         
>>>>>>           
>>>>>>             
>>>>> Of course, use ENUM or dbaliases.
>>>>>
>>>>>
>>>>> -- 
>>>>> Iñaki Baz Castillo
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>> !DSPAM:49bb9c46608611888415445!
>>>>>
>>>>>
>>>>>     
>>>>>       
>>>>>         
>>>>>           
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>>>>   
>>>>     
>>>>       
>>>>         
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>>>   
>>>     
>>>       
>>   
>>     
>
>
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