Simple trunk and outbound route - how??

OUtrixboxguy's picture

Hi,

I am using trixbox 2.8 and would like to know how to setup a simple trunk and outbound route? My network is setup as follows:

trixbox server --> Redfone-foneBRIDGE (T1/E1) <**** T1 Line ****> Norstar PBX.

Right now i have 24 channels available when i perform a dahdi show channels in the Asterisk CLI.

T1 = ESF, B8ZS, RBS

I am not sure if i should be adding a "Zap Trunk (DAHDI compatibility mode)" trunk, or a "Custom Trunk". If it is a custom trunk, what should my custom dial string be since i am going over a T1?

Lastly, which config files are written when i create the above? I would like to see if indeed the config files are being written.

Thanks.

p.s. i am a student trying to set this all up, so please bear with me :)


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We are helping you over at

SkykingOH's picture

We are helping you over at the trixbox web site.

Posting the same question over here is not going to get you any better results, simply a repeat of the process I am taking you through.

The student gets you less sympathy because you should read the documentation on context and then you would not have placed your trunks in a non-existant context.


I did not know that this

OUtrixboxguy's picture

I did not know that this site is dually monitored with the trixbox site. I apologize.

I am not asking for sympathy when i state i am a student, but rather patience.


Quote: I did not know that

SkykingOH's picture
Quote:
I did not know that this site is dually monitored

It is by me, however that is hardly official.

Since Fonality choose to take the FreePBX name off of trixbox and call it a fork it's not reasonable to expect the FreePBX folks to help.

It's very encouraging that the developers preview of FreePBX 3.0 is being rolled on an ISO. Maybe this is a sign of things to come or I am reading too much into it!

I support trix users for a variety of reasons, that does not mean I support the direction of the project. I use real FreePBX, support the FreePBX project and took the official OTTS training.

As a student of Voip you will learn that it is far more flexible to install your own Asterisk rather than be tied to a distribution. The distro's have also lowered the bar for entry into end user supported Voip and given the whole technology a black eye.

I don't care what anybody says it takes an experienced specialist to install and maintain an IP based phone system, open or closed architecture.