Displaying 20 results from an estimated 10000 matches similar to: "1.2.9.1 crashed today"
2007 Mar 19
4
Queue App - Free agent and waiting calls
<asterisk-users@lists.digium.com>Asterisk 1.4
I have strategy= leastrecent and autofill = yes
I have 2 agents, one is answering a call and the other is free and have some
calls waiting in the queue.
Only when the first agent hangup the second agent receive the first call in
the queue.
It happends some times.
This behavior still happend in 1.4.1 version.
Thanks a lot.
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2007 Jun 03
2
Chan_mobile issue
Hello,
I just did a fresh svn install of 1.4 trunk everything. Everything
compiles and installs just fine.
When I get to asterisk-addons, I cannot select chan_mobile in
menuselect.
Chan_mobile is not even an option in menuselect for asterisk trunk.
I tried the latest patch which failed in many places but did add an
option for chan_mobile in menuselect for asterisk but it still cannot be
2006 Mar 30
3
Please Help Test Quad PRI Using NFAS
Please help me test my setup by dialing 800.564.0215 and listen to the
queue for a bit. I have a quad port T1 with NFAS setup.
I can dial-out but I cannot dial any 800 numbers (Global Crossing says I
need LDS service and that will be a couple weeks) so I cant test it
myself. I need at least 24 callers to feel comfortable enough that it
is working properly.
Thanks,
Steve Totaro
2007 May 24
13
Bottom line on fax reception
So what is the bottom line? Does it work or not. I've heard stories it
works, it doesn't work, it kinda sorta works when it's not raining out side.
Everything under the rainbow.
What's the bottom line with recent updates on 1.2.x? Is it production ready
for fax? By production ready I mean that it just works all the time and
doesn't need any babysitting. Do I have to worry
2007 Mar 07
4
OT Vonage V-Phone Adapter (Possible Hack)
It would be cool to get one of these and see if it can be hacked and
loaded with your favorite SIP or IAX softphone. Looking at the pic, it
looks like the dongle is both a soundcard and memory stick. Heck, I
would be glad to have it if I could get the soundcard to work.
Might as well since it is free after rebate.
http://www.circuitcity.com/ssm/Accessories-for-Vonage-V-Phone-VPHONE/sem
2007 Apr 12
8
test
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii">
<TITLE>Message</TITLE>
<META content="MSHTML 6.00.2900.3059" name=GENERATOR></HEAD>
<BODY>
<DIV> </DIV></BODY></HTML>
2006 Jun 12
2
Hitting * in a queue call hangs up?
Can anyone explain why when I hit * (as in *2 to transfer) a call that
has come to me in a queue asterisk disconnects the call? All I have
to do is hit "*" and the call drops.
2007 Jun 06
3
1.4 Zaptel/Sangoma Issues on CentOS
Any ideas? Sangoma support is closed for the evening.
I have the latest Sangoma drivers and Asterisk 1.4 everything installed.
When I fire up asterisk, I keep getting "Primary D-Channel on span 1 up"
repeated over and over. The B channels never come up. There are no
errors in any of the logs, zttool, or the wanpipe tools.
Intense pri debug output:
< Unnumbered frame:
< SAPI:
2006 May 26
4
mpg123 or asterisk
should I use mpg123 with asterisk 1.2.7 or should i use the native
player asterisk has?
the target machine will receive heavy load.
also, has anyone succedded in compiling mpg123 in a dual core pentium
with centos 4.3 ?
--
-------------------------------------------
Erick Perez
Linux User 376588
http://counter.li.org/ (Get counted!!!)
Panama, Republic of Panama
2006 Apr 13
2
app_meetme.so
Hi all,
I'm using Asterisk 1.2.5 and , for some reason, when I install it, the
module app_meetme.so didn't install. Is there some way to download
that module, and add it to asterisk without re-install it?
Thanks in advance
Sebastian
2006 Feb 14
2
audio cuts out
has anyone experienced a problem where RTP audio cuts out when doing
30-40 concurrent channels via sip?
The box is freebsd 6, asterisk 1.2.4, voip only (no libpri, no zaptel -
not even a timing source)
The box has plenty of bandwidth, when a call to the same box is iax2 it
works, but when its sip a call gets connected a few frames of audio are
passed and then silence.
When the box is completly
2006 Jun 12
1
Single agent multiple queues....
Hi,
I have several agents, who all log into multiple queues.
What I want to happen (but doesn't seem to be) is:
Agent 5 is logged into queues 1,2,3
Agent 4 is logged into queues 1,3
A call comes into queue 1, and goes to agent 5.
Agent 5 answers the call and finishes it.
A call comes into queue 3.
I want this call to go to Agent 4, as opposed to going to agent 5
(which is what it is doing
2005 Feb 19
3
Still asterisk startup crash plz help
Hi,
First i would like to thank the kind people of the list who have
answered my previuos mail, but i am still stuck as asterisk still
crashes upon startup, i have read the install article at
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
and i have search the asterisk archives, but i still cant get asterisk
to work, i have tried reinstalling asterisk but it still complains and
2007 Apr 25
2
My Polycom IP 501 is formatted its file systemitself
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Noah Miller
> Sent: Wednesday, April 25, 2007 9:52 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] My Polycom IP 501 is formatted its file
> systemitself
>
> Hi Chandra -
>
>
2005 Jan 18
9
# Transfers.
So I have read and read and read... google is my friend and the wiki is
by brother...
However, I am still unclear on what the preferred method of using the
pound sign is.
If the Pound sign is set aside for Transfer.. Then when I make an
outbound call to my bank I can not "Enter my PIN followed by Pound"
Likewise if I turn off the ability to transfer when initiating a call,
my bank pin
2006 Nov 16
2
installing asterisk for Ubuntu Synaptic
I have an Ubuntu system and went into Synaptic and checked asterisk for
installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc
and got the following output with several errors and notices. Do I need to
do more or are these ok? I expected to have some conf files in
/etc/asterisk but there is nothing there.
Thanks!
Created by Mark Spencer <markster@digium.com>
2007 May 28
5
Blindside Web Conferencing
Hello,
We are creating a web-based conferencing application using Asterisk as the
voice conferencing server.
This as an open source project. We are trying to determine if there
is interest of the community and perhaps work together to improve the
application.
Using the web application, you can upload your powerpoint presentation,
manage the participants in the conference thru the web interface
2007 May 29
2
Agents.conf from realtime static
I am using Asterisk 1.4.4 on a CentOS 5 machine for a small call center
with 6 agents. I am using realtime for queues and sip and I am also
trying to use realtime static to load agents.conf. The only problem I
am having is that no agents are loaded when I start Asterisk. I have to
manually do a "module reload chan_agent.so" so the agents get loaded
from the database.
Obviously
2005 Aug 18
2
Updated Patch to chan_agent.c for PREACKANNOUNCE
First, many thanks to Greg Boehnlein for his patch to chan_agent.c
for adding a "preackannounce" option.
I am running CVS HEAD from 2005/07/31, and the patch failed in a
few hunks, since the code was refactored to add in some CASE
statements where there were compound if statements before.
Anyway, I have successfully updated the patch to work against head
as of 3 weeks ago, and would
2007 Apr 15
2
agents and music on hold with autoanswer..
My colleague left our company, then I have to manage all our phones
lines and asterisk: please, apologize me because I'm 'absolute
beginner' about voip/asterisk!!
Well... all seems work fine; we have some queues and some agents; the
"music on hold" works fine when the agent press the hold button on
the phone (thomson); the agents have the 'autoanwser' flag