Newbie: A simple asterisk bridge

Asterisk Asterisk Support
1

I’ve just setup an asterisk server (my first), and after a few days of struggle trying to work out all the functions, going through tutorials, and trying to take shortcuts using the gui, I think I’ve finally got my SIP’s registered. That said, I apologise if this has been covered somewhere else, but I’ve done searches but to be honest I’m not sure exactly what to search for.

What I’m trying to do is provide a connection from my Netcomm V100 to my VoIP provider KMoo, who appears to be using some odd VoIP equipment. The problem with directly connecting my ata with KMoo is I can make calls, but not recieve, and apparently my ata is incompatible with their stuff. So my plan here is to use an asterisk server to “bridge” the two if possible.

So far so good, the Netcomm ata registers with asterisk, and asterisk registers with KMoo (I think). Now my problem lies in setting up the extensions.conf file simply so when I dial a number it routs the call through KMoo, and when I recieve a call its routed to the phone (using the Netcomm ata).

What happens currently is I dial on the phone and it comes back with a hangup (not unexpected, as problem is in the dial function), and when I call the phone it says its not available (out of range or switched off).

My setup is as follows:
extensions.conf

[kmoo-in]
exten => s,1,Dial(SIP/home,25,r)

[from-user]
exten => s,1,Dial(SIP/kmoo,25,r)
exten => s,2,Hangup

sip.conf
[kmoo]
context = kmoo-out

[“phone number”]
context = kmoo-in

[home]
context = from-user

Any ideas on how I can pull this off?
I figure if I’ll try pulling this off before venturing too much further as this is all I need for the moment…

2

hi
as per your given dail plan & sip.conf. i think there is something missing in sip.conf settings . baecuse ur mentioed only single parameter.
also may be there is problem in extension.conf config. or context inclusion. plz check

3

[quote=“amit”]hi
as per your given dail plan & sip.conf. i think there is something missing in sip.conf settings . baecuse ur mentioed only single parameter.
also may be there is problem in extension.conf config. or context inclusion. plz check[/quote]

For reference I’ve only included settings which I thought were relevant- my provider has some specific settings to suit their system apparently (general,global, network, as well as contextual). That and the SIP settings appear to be working because the server registers with my provider ok, and I don’t get any more errors regarding auth failures. Hence I’ve only included the context variable so that you can see what I’ve entered there that relates to the extensions.conf.

My main concern is in the extension.conf with the settings I’ve input there. I’m not sure if I’m using the functions correctly.

What precisely do you mean by ‘there is a problem in extension.conf or context inclusion’? I’m a newbie on asterisk so I don’t know what problem there could be or I probably wouldn’t have asked here- I just don’t know what I’m checking for! I’m having trouble finding and getting my head around all the variables and functions in asterisk, so thats why I’m asking- plus I don’t know if I have the expressions correct.

If you do need to see the sip.conf I can post that too.

Thanks guys