It seems you have some problems with your fxo configuration. Can you post your zaptel.conf and zapata.conf?

I have configured Easyasterisk with a Digium TDM400 card with 2 FXS ports and 1 FXO port. I also have an SIP account with budgetphone and a VOIP phone (Grandstream)
I can call internal phones (2 analog phones and 1 VOIP phone), this is working fine. I can also dial out through the SIP account and my analog channel.

The problem is receiving a call. When I call the SIP account I directly get my voicemail (not from easyasterisk, but from budgetphone). When I call my analog line I get no answer (and no ringing phones).

I have tried filling in the phonenumbers in the DID list, but this is no soloution.

Strange thing is that when I view the CLI (Alt-F9) I can't see anything happening when a call comes in.
(You do see a lot of actions when you pick up the phone and dial a number.)

Can anybody help me to get Easyasterisk to answer my calls and put them through to my phones ??

Thanks.

Username and secret are available only on dynamic host configuration.
Go to each extension and set host on dynamic, change username and secret as for phone configuration.
On Linksys SPA 922, select Admin Login, Ext1, UserID and password in Subscriber information section.
Don't forget to reboot the phone.

Hello,

I've installed easyAsterisk and tried to register 2 SIP Linksys SPa922 phones. No matter how hard I've been tried the result was null. For some reason I am unable to set the "username" and "secret" parameters in "Local extension" section of web interface.
I'm out of ideeas. What is wrong? What can I do? So far I am unable to make a simple phone call, because the pshones doesn't register with the asterisk!
Any hints, advices, ideeas are highly welcome!
Thank you very much!

Hi all, I'm stuck with how to pass the return value to asterisk. That is, a value has to be passed from Perl script to asterisk. The variable returned in Perl has to be passed to asterisk, where if the return value is 1 from Perl , i have to do something and if return value is 0, i have to do something else. I have written code like this.
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sujith

easyAsterisk does not include A2billing integration. You can install it manually including sip and iax configuration files used by A2billing from "General Settings => SIP/IAX2 Protocol => Advanced" menu.

I would like to know is Easyasterisk works with A2billing please advice thanks.

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Hello,

At the moment mfcr2 support is not planned. We are currently working to update easyAsterisk to Centos 5 and asterisk 1.4.

Will easyasterisk include mfcr2 for unicall?

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You need to configure Operator Panel from "General Settings => Operator Panel Management" menu.

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You can add a new local extension from "Pbx Seggings" menu and choose IAX2 as protocol used by the new extension (default is SIP).