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.

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

I can call and be called by the analog number.
The VOIP account can call out but can not be called.
(On the phone I can place and receive calls when I use the direct outsite line). Only when I use EasyAsterisk to get called the phones will not ring (I put all the phones in a group to receive calls and during test I disable the direct line on the phone)
When I call the VOIP number, I directly get my (budgetphone) voicemail.

Thanks in advance.

FYI: (created by the system)

zaptel.conf:

loadzone=nl
defaultzone=nl

fxoks=1
fxoks=2
fxsks=3

zapata.conf:

[channels]
language=en
immediate=no
faxdetect=no

; ADAPTER 1: ANALOG port 1 FXS
signalling = fxo_ks
callerid=100 <100>

group =
context = 100
echocancel = yes
echocancelwhenbridged = yes
rxgain = 0.0
txgain = 0.0
echotraining = 400

channel => 1

; ADAPTER 1: ANALOG port 2 FXS
signalling = fxo_ks
callerid=101 <101>

group =
context = 101
echocancel = yes
echocancelwhenbridged = yes
rxgain = 0.0
txgain = 0.0
echotraining = 400

channel => 2

; ADAPTER 1: ANALOG port 3 FXO

usecallerid = yes
callerid = asreceived
busydetect = yes
busycount = 4
signalling = fxs_ks
context = incoming
group = 1
echocancel = yes
echocancelwhenbridged = yes
rxgain = 0.0
txgain = 0.0
echotraining = 400
channel => 3

Quote:

Please check your sip trunk configuration. You need to set the "type" as "friend" and enable registration. If you execute "sip show registry" in asterisk console you can verify if the server is correctly registered to budgetphone. Remember also to set correctly "Attendant Console".