Hi admin,
Telco advised NT mode and Point to Point.

Error:
[root@Econet ~]# asterisk -r
Asterisk 1.4.19.2, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.4.19.2 currently running on Econet (pid = 26748)
Verbosity is at least 3
P[ 1] !!! Could not Get the L2 up after 3 Attemps!!!
P[ 1] !!! Could not Get the L2 up after 3 Attemps!!!
P[ 1] !!! Could not Get the L2 up after 3 Attemps!!!
P[ 1] !!! Could not Get the L2 up after 3 Attemps!!!
P[ 1] !!! Could not Get the L2 up after 3 Attemps!!!
P[ 1] !!! Could not Get the L2 up after 3 Attemps!!!
Econet*CLI>


Please advised

Regards
Nazrul

Quote:

Please check if the configuration of your ISDN adapter (point to point / point to multipoint) meets che configuration of your BRI Line (your telephon company should know this).

Quote:

Login in console as root and enter in asterisk console typing:
asterisk -r

Please post your console log.

Hi admin,
I have follow as your advised.now i can make local extension to extension call.
i am unable to call out from extension to mobile through ISDN BRI Line.

Niether incoming call on the number which provided by Telecom.
I have some screenshoot,i have no idea how to upload for you to View.

what is the procedure to see call hit from Bri line for incoming call?

Waiting for your reply.

Regards
Nazrul

Hello, you can proceed as follows:

- Configure the B410P (Hardware Settings => mISDN Hardware)

- Configure ISDN Trunks (Trunks Management => mISDN Trunks)

- Configure additional SIP/IAX/H323 trunks if necessary

- Configure your local phone extensions (PBX Settings => Local Extensions)

- Create your outbound routing rules and assign rights to local phone extensions (Calls Management => Outbound Routing)

- Create queues, meetme, IVR, ecc. if necessary

- Configure incoming calls rules (Calls Management => Attendant Console + Direct Inward Dial (DID))

You can find configuration details with examples in easyAsterisk User's Manual.

Hi Everybody, I am new to easyasterisk and 1st time installed into my IBM server with Digium B410P. Everything goes smoothly and I can access the server via Web http://192.168.2.X/astconf/ with admin/admin login successfully all options can seen. Now please advise me all steps Sequence for Configuration Easyasterisk server as following requirement. This for my Company customer care Hotline 4 extensions with 4 IP Phone Incoming Call Number 26112262 (Telco provided) 1) Customer Call( 4 concurrent call)->26112262-->(Digium B410P )Easy Asterisk Server -->Extensions 2) All extensions should able to call out to any mobile or land line through SIP/H323 Trunks. Please advised as soon as possible. Will be highly appreciated. Thank you Nazrul Nazrul_70@yahoo.com.sg

Quote:

Hello,
Trixbox integrate freepbx while easyAsterisk use a proprietary web interface. easyAsterisk web gui is very easy to use. You can try both and choose what you like.

nobody used trixbox ?

I will setup a voip server .
trixbox and easyasterisk which point better ?

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".

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