Inbound call routing..

wipeout

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#1
Hi,

Sorry if this is a common question..

I have just setup Elastix for the first time and have managed to setup my FXS and FXO channels but I am having a problem routing inbound calls from my PSTN line..

I have setup a DID for channel 1 on the Zap channel DIDs page..

If I setup an inbound route for ANY/ANY then I can get my SIP phone to ring when a call comes in but if I setup the inbound route to <myDID>/ANY then I get the message that the number is not in service..

Is there a trick to getting this to work?

Thanks..
 

Patrick_elx

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#2
In unembedded freepbx did you set the DID for your channel in the Zap Channel DID tab?

In your inbound route did you set the exact same DID?
 

DaveD

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#3
You need to change in /etc/asterisk dahdi_channels.conf to read context=from-zaptel instead of context=from-pstn
Also do the same for chan_dahdi.conf

Easiest way to do this if you are using a windows box is grab winscp and edit the file
 

wipeout

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#4
DaveD said:
You need to change in /etc/asterisk dahdi_channels.conf to read context=from-zaptel instead of context=from-pstn
Also do the same for chan_dahdi.conf

Easiest way to do this if you are using a windows box is grab winscp and edit the file
That solved it.. :)

Thanks..
 

byaru1

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#5
wipeout said:
DaveD said:
You need to change in /etc/asterisk dahdi_channels.conf to read context=from-zaptel instead of context=from-pstn
Also do the same for chan_dahdi.conf

Easiest way to do this if you are using a windows box is grab winscp and edit the file
That solved it.. :)

Thanks..
That worked for me too!

eb
 

Furry

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#6
Done a recent yum update, and version is now 1.5.2-2.3.

I think this update must have cause the 'all circuits are busy now' problem.

I've tried this (and several other things, from other threads), but I can't get rid of this - it's driving me mad!

Is there a *definitive* explanation / fix for this?

TIA,
Dave.
 

Patrick_elx

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#7
you should probably have opened a new thread for that.

Check the status of your extensions. Are they online?
can you call from one extension to another?

if not try the following:

Check that your unembedded freepbx is updated (including all modules).

Still in freepbx, go in each extension edit one of the setting to the same value and click save.
Do the same for each trunk and ring group.
eventually click apply and try again to see if it's working.
 

Furry

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#8
Patrick_elx

Yes, you're right, I should have started a new thread - I believed I had posted to another thread entirely, but somehow posted to this one. Sorry about that.

Thanks for your suggestions. I actually found that setting the Trunk identifier back to the default 'g0', rather than '1' (i.e. Zap/g0, not Zap/1) fixed this. I don't know why.

Thanks,
Dave.
 

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