[OpenSIPS-Users] Opensips + asterisk 1.4

[Willian Mazzardo - SYSSVOIP]

Hi Nick ...

I have tried with your modparam set but no look. If I register with dns
name, the register not work.

What I have? maybe help you canhelp me with what I need ;)

I have Asterisk to handle registration, routes and billing, but my customer
base are increasing, and I want use Opensips to handle REGISTER and route
PSTN calls to one of my asterisk box. (load balance) but this is later.

My customers have Asterisk, Linksys PAP2 adapters and sofpthones, and all
of them are registering with my asterisk. Some with IP Address, others with
one of my dns names.

I want to implement IM, Video calls, and presence to offer to my customers
something different from others Voip company.

I expect use opensips to call SIP > SIP customers directly, without
asterisk using rtpproxy when its needed,  and PSTN calls routing to
asterisk until I found some billing system who I can comprehend and know
how to use.

Thats all ;)



Willian Mazzardo
Depto TI - SYSSVOIP
www.syssvoip.com.br
55 3537 2030


2013/7/18 Nick Khamis <symack at gmail.com>

> There is also:
>
>
>    1. modparam("auth_db|usrloc|uri", "use_domain", 1)
>
> Please change that to 0. It's been a while since I have dealt with
> REGISTER authentiacation issues. Are you sure
> you need it? It's quite a resourceful process as the number of clients
> increase. What we do now, is use:
>
> 1) The address table
> 2) Dialplan
> 3) Dynamic Routing
> 4) IPTables
>
> To enforce who's INVITE gets processed by our servers. No registration
> required.
>
> If you really want to handle REGISTER, I will take a closer look. Until
> then maybe look at Chapter 5 (Page 90),
> of
> ftp://115.146.120.141/voIP/Building%20Telephony%20Systems%20with%20OpenSIPS%201.6.pdf.
> I know the
> answer is in there because I dealt with your issue a long time ago.
>
> Kind Regards,
>
> Nick
>
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>
>
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