connect SIP Trunk to Brekeke SIP Server

1. Brekeke Product Name and Version:
Brekeke SIP Server 2.4.3.0/286

2. Java version:
-

3. OS type and the version:
Windows XP

4. UA (phone), gateway or other hardware/software involved:
External VOIP Provider

5. Your problem:

Hello,

We have currently a working Setup internally which runs over PSTN but we want to switch from PSTN to SIP. Now we finally received a test account from the new SIP Provider which will be in the common month and years our Main Telephony Provider.

So the actual Setup is:

PSTN Gateway - PBX - SIP Brekek Server - Client


Now our PBX Server Supporter wanted me to connect the SIP Trunk to the Brekeke SIP Server to test the connections between Client and the new Provider. And this is where i struggle, i tried some rules with a upper registration but it does not work.

So my Question how can i connect a SIP Trunk to the Breke SIP Server?

I received the following Datas:

User: 43720***
Passwort: ****
Registrar: 12.34.555.666
NAT: yes
Codec: alaw, g729
Channels: 2
Ports: 5060- 5080

So: i need to setup a outbound call which routes from the Client over the Brekeke SIP Server to the new provider Registrar.


Thank you very much for you help!


Br,

Dom

Well its a basic Question, forget the Rest:

How can i connect a SIP Trunk to the SIP Server? And do i need to setup any special rules to let inbound and outbound calls work?

Thanks!

Guys please help me with this one, i need to Set this up soon.

Thanks!

You didn't provide enough information yet...

Which PBX product are you using?
How do you plan to connect a SIP Trunk?
Is it behind a PBX as the PSTN gateway?

Also I recommend you update Brekeke SIP server to the latest version.
The ver2.4.3.0 was released 5 years ago. It means very old.

Genom88 wrote:Guys please help me with this one, i need to Set this up soon.

Thanks!

Genom88

Hey what's up! I see what you're talking about. What you should do is actually pass the traffic outbound from the extension from your pbx over to the Brekeke Sip Server. From there, I recommend you try out THRU REGISTRATION and mind you not if you DONT see the REGISTRATION show up on the Admin CP of the ITSP you are using.

I have tried such with great success over a variety of ITSP [internet Telephony Service Provider] and ask you try this. Who is your provider? What is the hard/soft ware of your PBX? Please reveal such and we can even give you better insight. I've personally discovered the magic of Brekeke Sip Server would be using the THRU / UPPER REGISTRATION.

Check the Wiki on such topic as well for further details. Once given these details we can assist further. Also, Version 3 is ALOT better as well. I would upgrade if you can do such. You get extra tabs to separate production rules from brekeke rules, etc that's a huge advance alone worth upgrading.

Thanks,
Kent C.