[OpenSIPS-Users] Opensips + asterisk 1.4

[Willian Mazzardo - SYSSVOIP]

Nick ... i`m  making some tests, and I changed this block in my
opensips.cfg and WORKED with any domain I define.

                # authenticate the REGISTER requests
                if (!www_authorize("10.1.1.2", "subscriber"))
                {
                        www_challenge("10.1.1.2", "0");
                        exit;
                }

                if (!db_check_to())
                {
                        sl_send_reply("403","Forbidden auth ID");
                        exit;
                }


if I set into softphone the  DNS name in domain field, work, and if I
define 10.1.1.2 also work.

;)

I think this is what I was looking for.



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


2013/7/18 Willian Mazzardo - SYSSVOIP <willian at syssvoip.com.br>

> 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
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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