Re: [Users] Getting Started - Basic Proxy

Hello,

On 03/10/06 15:49, Roberto Pereyra wrote:

Hi

Where I write the sip server (pstn gateway) username and password ?

non-rfc compliant authentication can be achieved using the uac module. See the readme of the uac module for more details. http://openser.org/docs/modules/1.1.x/uac.html

Or the gateway must allow per IP or host logins ?

This is the recommendable way at this moment.

Cheers, Daniel

roberto

2006/3/10, Glenn Dalgliesh <glenn@routerboy.com mailto:glenn@routerboy.com>:

You most likely need to add a section in the correct location in
the cfg to change the dst based on the dialed digits such as
below. You need to keep in mind where you put this in the script.
Keep in mind that you don't want to step on the lookup for locally
registered contacts.


                if (uri=~"^sip:1[0-9]{10}@") ||
(uri=~"^sip:[0-9]{10}@") || (uri=~"^sip:011[0-9]+@.*") {
                        if (uri=~"^sip:[0-9]{10}@") {
                                prefix("1"); # if sip provider
require leading 1
                        };
                        rewritehostport("192.168.1.1:5060
<http://192.168.1.1:5060>"); # to you sip provider

                        route(1); # send to t_relay route section
                        return;
                };


    ----- Original Message -----
    *From:* Robert Kennedy <mailto:rob.s.kennedy@gmail.com>
    *To:* users@openser.org <mailto:users@openser.org>
    *Sent:* Friday, March 10, 2006 7:29 AM
    *Subject:* [Users] Getting Started - Basic Proxy

    Im currently learning OpenSer and have accomplished the following:

    Installed: Centos 4.2
    OpenSer: V1.0.1

    I have 2 x-lite client registering and calling each other.


    Question: How do I proxy on SIP requests to a 3rd part SIP
    Provider. eg. Sipgate

    xlite --->  Ser ---> Sipgate

    Im just getting confused on how to configure the next hop routing?

    Currently as expected im getting a 404 Not Found unless I call
    one of the x-lite phones. I want to be able to terminate out
    to a SIP-to-PSTN provider.

    Regards

    Rob.

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-- Ing. Roberto Pereyra ContenidosOnline Servidores BSD, Solaris y Linux Soporte técnico ISPs Jabber ID: rpereyra@lugmen.org.ar mailto:rpereyra@lugmen.org.ar

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