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? > [email protected] 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: [email protected] [mailto:[email protected]] > Sent: Tuesday, April 21, 2009 11:51 AM > To: [email protected] > Cc: [email protected]; [email protected] > 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: [email protected] [mailto:[email protected]] >> Sent: Tuesday, April 21, 2009 11:38 AM >> To: [email protected] >> Cc: [email protected]; [email protected] >> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab >> >> Hi Robert, >> >> you mean, at the end, you want to sent it to [email protected] phone, but >> with [email protected] 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: >>> >>> [email protected] >>> [email protected] >>> [email protected] -- forward to ---> [email protected] >>> ... >>> [email protected] >>> >>> Another thing I'm currently completely lost in is how to handle the >>> outgoing part. [email protected] should be able to place outgoing calls with >>> the originator set to one of the [email protected] ... [email protected] >>> addresses. >>> >>> How can I achieve this behaviour? Thanks a lot! >>> >>> >>> Regards, >>> Robert >>> >>> >>> -----Original Message----- >>> From: [email protected] >>> [mailto:[email protected]] On Behalf Of Bogdan-Andrei Iancu >>> Sent: Monday, March 16, 2009 9:14 AM >>> To: [email protected] >>> Cc: [email protected] >>> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab >>> >>> Hi Elnor, >>> >>> [email protected] 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: >>> d...@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" <[email protected]> >>>> To: <[email protected]> >>>> Sent: Saturday, March 14, 2009 3:59 PM >>>> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab >>>> >>>> >>>> >>>> >>>> >>>> >>>>> El Sábado, 14 de Marzo de 2009, [email protected] 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 >>>>> [email protected] >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> !DSPAM:49bb9c46608611888415445! >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Users mailing list >>>> [email protected] >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> >>> _______________________________________________ >>> Users mailing list >>> [email protected] >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> >> >> > > > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
