a better way is to to load the driver with all spans set to E1 by running
modprobe wcte11xp t1e1override=15
or edit wcte11xp.c and change 'static int t1e1override = -1;' to 'static int
t1e1override = 15;'
Diyanat
From: Robert Andersson [EMAIL PROTECTED]
Reply-To: Asterisk Users
well roughly 80 calls on g729 or 120 on g711, figures may differ in
realtime, 100 gb bandwidth may not be sufficient, you will have to know the
actual throughput too
you should check this tool for bandwidth calculation
http://www.asteriskguru.com/bandwidth_calculator.php
Diyanat
From:
add to iax.conf on server1
register = username:[EMAIL PROTECTED]
on server1
lets say extension 1001 on server1 will transfer the call to extension 1002
on server2
exten = 1001,1,Dial(IAX2/[EMAIL PROTECTED]) ; replace server2 with ip/domain of
server2
on server 2 extension 1002 will join a
You may enable polarity reversal on the line, ask your telco about it
then add the following to zapata.conf
hanguponpolarityswitch=yes
answeronpolarityswitch=yes
you can also use a call progress detector such as
http://www.broadcastboxes.com/products/CP-2_lit.html
Diyanat
From: Olivier
You can use ChanSpy() module to monitor
Diyanat
From: Rajkumar S [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Errors-To: [EMAIL PROTECTED]
Return-Path: [EMAIL PROTECTED]
X-OriginalArrivalTime: 08 Jan 2006 19:40:13.0831
Hi
The new meetme i feature in asterisk1.2.1 for annoucing user join/leave
is good, but the initial steps to record the name and confirm seems lenghty,
the user shoudl just say the name and get into the conference, How can i
disable the confirmation of the name recorded before entering the
Hello!
I am using asterisk 1.2.1 with a digium TDM card and trying to reduce the
dialout delay to 1/2 secs at the most, i could bring it down from 6/7
seconds to 3/4 seconds by tweaking the config and tone/zone/dtmf settings in
the source, still this is not acceptable as the regular pstn
Hello!
I have a TE410P quad span card with 4 E1, i am using asterisk 1.2.1, i was
using it without any issues earlier with just 1 E1 on span 1 and i recently
plugged in 3 more E1's, only span 1 is working, the e1 for span 1 is from a
different provider then the rest, the settings are same for
i am using asterisk 1.2.1 with mysql 5 without any issues, please check your
configuration again, make sure you have hostname=localhost too and the
dbname, user, password are correct
[global]
hostname=localhost
dbname=databasename
user=user
password=password
port=3306
Do you have 't' or 'T' in the Dial Application?
Diyanat
From: Douglas Garstang [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
check http://www.asteriskguru.com/tutorials/
Diyanat
From: sukrit [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] getting started
Date: Fri, 16 Dec 2005 09:06:28
in the sip.conf have the following enteries
; for regsitering with ser
register:seruser:[EMAIL PROTECTED]:5060;(put ser machine ip:port)
;add a user for the ser machine
[seruser]
type=friend
host=0.0.0.0 ;(put ser machine ip here)
nat=no ;(change as needed )
canreinvite=yes ;(change as needed)
of
defining it on extensions.conf... Would it be the same in extensions.conf?
Should I write $AGI-exec('Dial', 'SIP/[EMAIL PROTECTED]'); to dial it from
AGI script (perl), is this correct?
Thank you very much,
Ryan
At 01:45 PM 12/16/05, Diyanat Ali wrote:
in the sip.conf have the following
in queues.conf
change
joinempty = no
leavewhenempty = no
to
joinempty = strict
leavewhenempty = strict
From: Xavier Gil [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject:
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