If your issue is with Asterisk then you should post to that list not the SER list.

As long as you have the context correct and you have the autocreatepeer=yes correct then it should work.

Dave

On Monday 11 October 2004 14:20, Bastian Schern wrote:

Hi Dave,

in my case this will not work. The Asterisk-Server is called "parrot" and the SIP-Server "lion". It looks like the forwarding from SER to Asterisk is partly working, but Asteris will reject the call. On my Asterisk I got this message: Oct 11 15:15:08 NOTICE[147465]: chan_sip.c:6978 handle_request: Failed to authenticate user "Bastian" sip:bastian@lion;tag=4omawlzf54

What is wrong?

David Simmons schrieb:

...

use the default cfg to start then put this in:

    if (method=="INVITE") {
# PSTN destinations begin with a 9.
            if (uri =~ "sip:9[0-9]+@*"){
	rewritehostport("192.168.0.50:5060");
	t_relay();
                    break;
            }
    }

if they dial 9 then a number they will go to the Asterisk server on 192.168.0.50.

Put this in sip.conf

[general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls disallow=all allow=ulaw autocreatepeer=yes

then in the extensions.conf:

[sip] ; this is where redirects from the proxy wil come - both pstn & voicemail ignorepat => 0 exten => _9X.,1,StripMSD,1 exten => _X.,2,SetCallerID,922 exten => _X.,3,Dial(Zap/1/${EXTEN},40,tr) exten => _X.,4,Hangup

The above will vary for what hardware you have in your box.

Obviously this is a failry crude example but should get you going.

Dave

On Monday 11 October 2004 13:50, Bastian Schern wrote:

...

Hi,

since a while I try get Asterisk and SER work together. But until now I have no success. I want to use Asterisk as Gateway to the old telephone world. Is there somebody who can give me a small example of the ser.cfg and the Asterisk config files.

This will be very nice.

Thanks Bastian

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