[OpenSIPS-Users] Multi-Domain Setup with PSTN Connectivity

Robert Borz robert.borz at web.de
Wed Jan 21 09:55:14 CET 2009 

Hmm, we're trying to handle everything in SER what's possible, having a central point of configuration. Asterisk just translates 1:1 between the PSTN and SER.

For accounting we want to use FreeRADIUS. I'm aware of the problem of lost BYEs in a dialog which leads in an undeterminable call duration. To overcome this, I want to use the session timers introduced with SIP and SER. Do you have any experience using them? I'm just asking to get some more detailed information and maybe you choose asterisk for accounting because the former didn't work well... ;)


-----Original Message-----
From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Mark Sayer
Sent: Tuesday, January 20, 2009 9:44 PM
To: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] Multi-Domain Setup with PSTN Connectivity

It was one way to make sure that we could accurately account for call 
times and provide all calls with PBX capability. I'm sure it's not 
the only way but it works well for us.

Mark

At 09:16 a.m. 21/01/2009, you wrote:
>Hmm, sounds interesting. If I understood it correctly, it looks 
>something like this for calls from the PSTN:
>
>---inbound pstn-->[OpenSER]--->[Asterisk]
>                                     |
>[UA]<----------[OpenSER]<-----------+
>
>Why did you choose _not_ to handle such calls directly within SER 
>without involving the Asterisk machine, here?
>
>
>Now I set the "use_domain" parameter to 0 for all SER modules used, 
>and everything works fine. But I'm not sure if this is really what I want...
>
>
>Robert
>
>-----Original Message-----
>From: users-bounces at lists.opensips.org 
>[mailto:users-bounces at lists.opensips.org] On Behalf Of Mark Sayer
>Sent: Tuesday, January 20, 2009 9:05 PM
>To: users at lists.opensips.org
>Subject: Re: [OpenSIPS-Users] Multi-Domain Setup with PSTN Connectivity
>
>We successfully run things a bit differently. Our in-bound from pstn
>goes to OpenSER (and may be translated to an Asterisk extension using
>dbalias) then to Asterisk. If the final destination is a UA then it
>goes back out thru OpenSER. Out-bound to pstn goes directly out from
>Asterisk. OpenSER handles all registration and NAT. Works well in
>production and config for both OpenSER and Asterisk is relatively
>simple and easy to maintain.
>
>Mark
>
>At 08:19 a.m. 21/01/2009, you wrote:
> >Ok, just hacking around I succeeded with the following code snippet
> >to rewrite the domain part:
> >
> >avp_db_query("SELECT domain FROM uri WHERE uri_user='$rU'",
> >"$avp(i:678)");                avp_pushto("$ru/domain", "$avp(i:678)");
> >
> >And the IP address of the SER server in the INVITE gets replaced by
> >the domain and everything works well. But I can't imagine that this
> >is what people do in a situation like mine...
> >
> >
> >-----Original Message-----
> >From: users-bounces at lists.opensips.org
> >[mailto:users-bounces at lists.opensips.org] On Behalf Of Robert Borz
> >Sent: Tuesday, January 20, 2009 7:23 PM
> >To: users at lists.opensips.org
> >Subject: [OpenSIPS-Users] Multi-Domain Setup with PSTN Connectivity
> >
> >Hi,
> >
> >I'm currently setting up OpenSIPS/OpenSER with Asterisk as a PSTN
> >gateway. As starting point I'm using the configuration wizard from
> >sip:wise [1].
> >
> >After some modifications to the module configurations it works fine,
> >without any (obvious) failures.
> >
> >Both, SER and Asterisk has public IP addresses on their its
> >interfaces and SER is setup for multiple domains. Multi-Domain
> >support also works fine for calls handled within SER, and also does
> >PSTN termination when forwarding non-local calls to the right domain
> >(sip proxy) and non-local-pstn-calls to the Asterisk machine.
> >
> >The problem occurs by receiving calls from the PSTN, from the
> >Asterisk machine:
> >
> >- Incoming Call from the PSTN to 555123456 gets forwarded from the Asterisk
> >   machine to the SER.
> >
> >- The INVITE is from sip:555123456@<IP asterisk> to
> >   sip:555123456@<IP openser>
> >
> >- And the uri table in the database backend looks like this:
> >
> >    id | username  |   domain   |   uri_user   |       last_modified
> >   ----+-----------+------------+--------------+----------------------------
> >     1 | user10000 | domain1.de | 555123456    | 2008-12-26 15:32:18.761647
> >     4 | user10001 | domain1.de | 555123457    | 2008-12-29 20:47:11.740234
> >     5 | user10002 | domain2.de | 555123458    | 2008-12-30 10:46:36.455437
> >
> >Well, for the uri_db module the use_domain parameter is disabled
> >(set to 0), so the is_uri_local() function consideres the call as a
> >local call (that's right). But now trying to lookup the location
> >table the user is not seen as online. The user for 555123456 is
> >registered as 555123456 at domain1.de.
> >
> >My uri_user's are unique and are equal to the PSTN numbers
> >associated with the users (including the area prefix). Users from
> >different domains hosted by the same SER gets PSTN calls from the
> >same Asterisk server.
> >
> >What's the usual way to lookup the domain part and rewrite the
> >INVITE so SER can relay the call to the correct domain and the user
> >isn't seen offline/not registered by the lookup("location") method?
> >
> >I can't imagine to be the first facing this problem. ;)
> >
> >Any help or suggestions are appreciated...
> >
> >
> >
> >Bye,
> >Robert
> >
> >
> >[1] http://www.sipwise.com/index.php/products?start=3
> >
> >
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> >
> >
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>
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