Urgent Help needed Regarding Problems with Outgoing Calls in Elastix

Dear Members,

First of all , thank you for creating such a wonderful community.

This is my first post in the forum. As this site is regarded as one of the best in VOIP world so i decided to join and become part of such a wonderful community.

Dear geeks and gurus i have a problem which i am unable to solve. I am new to Asterisk world , so decided to go with Elastix.

My Scenario is as under:

I have four port E1 Card and i have connected Elastix with Alacatel PBX , after configuring it as per the description in this site i came accross with the following problem.

I can call from Alcatel phones (connected with Alcatel Pbx) to Elastix but not vice versa, that is i am unable to call from my sip phone (a softphone i.e) to Alcatel phone and it gives the error " All Circuits are busy now, please call again later" and a error message displayed on Softphone is " Call Declined Error 603".
Calls from Alcatel pbx to elastix are smooth and without any problem.

Any help in this regard will be highly appreciated.

Thanks & Best Regards

Just added the Dial Plan Manually for extensions mentioning dahdi.

Thanks all for your replies and concern.

Finally i have solved the problem, the problem was in extensions.conf file.

Any one having “All Circuits are busy at the moment” and using E1 Cards can contact me if they need any help and i will try my utmost to solve their issue.

extensions.conf is owned by FreePBX and should not need to be modified, can you detail what you did?

Since the call is over E1 there’s no need for SIP i assume.

Do you have outbound routes defined?

It could also be the groupings you do in your chan_dahdi.conf or dahdi-channels.conf that will then correspond to your trunk setup in FreePBX and corresponds to your outbound routes…

Otherwise, place a call and paste your #asterisk -rvvvv logs here…

hi darklord,
did you configure a sip trunk?
Have you already configured a outbound route to the trunk?
If you have configured a sip trunk, what does “sip show peers” show when you enter it on the Asterisk CLI?