[OpenSIPS-Users] Rewrite R-URI for incoming PSTN calls

Mon Aug 10 21:23:32 CEST 2009

Nevermind my last post.  I hadn't worked with the PDT MI functions and didn't
know about OpenSIPS reading the database at startup and then needing to use
the MI functions to add or edit stuff in realtime.

Read the following on the PDT page and that should be the answer to my
question "The database is loaded by OpenSIPS at start up time. The module
uses only the cache to look up domains. If you want to add or delete a new
prefix-domain pair at runtime you have to use MI FIFO commands. All changes
made via these commands are applied to database and the cache is updated
correspondingly."

osiris123d wrote:
> 
> I was wondering how to solve this issue and sure thing using PDT and
> inserting the info into the PDT table worked, but I noticed that after I
> insert data into PDT via mysql I have to restart OpenSIPS in order for me
> to be able to call the DID number.  Does this make sense?  I figured I
> wouldn't need to restart OpenSIPS.
> 
> 
> Andreas Westermaier wrote:
>> 
>> Hi Robert,
>> 
>> that's almost the same issue we ran into some day... the best what you
>> can do (maybe for future setups) is what Bogdan already told you.
>> 
>> One solution would be using the pdt module. It allows you to rewrite the
>> domain part and do matching against a given prefix (here the DID).
>> 
>> In your example you could place the DIDs in the pdt table and associate
>> them with the destination domain for which they should be available. E.g.
>> 
>>  sdomain |    prefix    |   domain    
>> ---------+--------------+--------------
>>  *       |        10000 | example1.com
>> 
>> Using the prefix2domain("2") function in your opensips.cfg leaves the
>> prefix untouched and rewrites only the ruri's domain part:
>> 
>> 10000@<ip-address> ---> 10000 at example1.com
>> 
>> Now the lookup function should be able to determine the right callee for
>> the DID 10000.
>> 
>> 
>> Regards,
>> Andreas
>> 
>> 
>> -------- Original-Nachricht --------
>>> Datum: Mon, 01 Jun 2009 14:14:48 +0300
>>> Von: Bogdan-Andrei Iancu <bogdan at voice-system.ro>
>>> An: Robert Borz <robert.borz at web.de>
>>> CC: users at lists.opensips.org
>>> Betreff: Re: [OpenSIPS-Users] Rewrite R-URI for incoming PSTN calls
>> 
>>> Hi Robert,
>>> 
>>> I suggest a kind of separation between SIP ids (which are multidomain 
>>> -userpart may appear in more domains-  and the identifier is 
>>> user at domain) and the DID (or numbers that are unique and do not belong 
>>> to any domain).
>>> 
>>> This will solve the problem of SIP IDS with domain and DIDs without
>>> domains.
>>> 
>>> To map DIDs over the SIP accounts, use aliases (aliasesdb).
>>> 
>>> Regards,
>>> Bogdan
>>> 
>>> Robert Borz wrote:
>>> > Hi,
>>> >
>>> > While now running our sip proxy quite a while our requirements changed
>>> and we now need a multidomain setup with pstn connectivity, which almost
>>> works already. But I need help for solving a special issue... ok, here's
>>> the
>>> problem...
>>> >
>>> > Imagine we got the accounts 10000 at example1.com and 20000 at example2.com
>>> (both got pstn-numbers as their uri parts).
>>> >
>>> > In our current configuration it is possible to dial pstn numbers by
>>> omitting the domain part (use_domain=0, usrloc) and every body can reach
>>> the
>>> other (e.g. 1000 at example1.com just dials 20000 and ends up by talking to
>>> 20000 at example2.com). It is also possible to receive pstn calls from our
>>> gateway
>>> which addresses the users by an ruri like "10000@<ip-address>".
>>> Everythings great.
>>> >
>>> > But now we set use_domain=1 (usrloc), because we want to allow same
>>> uri-parts for different domains (e.g. userxy at example1.com and
>>> userxy at example2.com).
>>> >
>>> > If now 10000 at example1.com dials a pstn number, let's say 12345, and
>>> omits the domain part, the sip server redirects the call to our
>>> pstn-gateway
>>> and the call get's established.
>>> >
>>> > But if we got the accounts 10000 at example1.com and 10000 at example2.com
>>> where only the first is the user allowed to receive pstn calls for the
>>> pstn
>>> number 10000 and the second is not, the call from our pstn gateway comes
>>> in
>>> with ruri=10000@<ip-address> and because use_domain=1 is set for the
>>> usrloc
>>> module, the sip server cannot direct the call to the user
>>> 10000 at example1.com because 10000 at example1.com is not the same domain as
>>> 10000@<ip-address>.
>>> >
>>> > What we would need is an additional mapping/translation. On an
>>> incoming
>>> call from our pstn gateway for 10000@<ip-address> must be translated for
>>> either 10000 at example1.com or 10000 at example2.com. So we need to rewrite
>>> the
>>> ruri. The best would be a database table holding at least the following
>>> information:
>>> >
>>> > uri, dst_domain
>>> > 10000, example1.com
>>> > 20000, example2.com
>>> >
>>> > So if we receive a call from our pstn gateway, we can lookup the
>>> destination domain from this table an rewrite the ruri accordingly.
>>> >
>>> > Is there already a module/method for achieving this goal?
>>> >
>>> > This is a lot of text and I hope you can follow me... any help would
>>> be
>>> really appreciated.
>>> >
>>> >
>>> > Regards,
>>> > Robert
>>> > ____________________________________________________________
>>> > Text: GRATIS für alle WEB.DE-Nutzer: Die maxdome Movie-FLAT!
>>> > Jetzt freischalten unter http://movieflat.web.de
>>> >
>>> >
>>> > _______________________________________________
>>> > Users mailing list
>>> > Users at lists.opensips.org
>>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>> >
>>> >   
>>> 
>>> 
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
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>> 
>> 
> 
> 

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