Displaying 20 results from an estimated 1000 matches similar to: "Manager connection problems"
2006 Feb 28
2
Conference bridge dimensioning
We are using an Asterisk box to do conferencing right now. I have had
about sixteen active lines in conference and the quality was acceptable.
We now have a need for 50 people to conference at one time. Does anyone
have enough experience doing this to give me some pointers. Will it even
be reasonable to try this? Is the mixing done on the the hardware, I
plan on using a quad span t-1 card from
2006 Mar 17
2
choppy recorded sounds in asterisk
I have installed asterisk on numerous servers. Every install was done on
Fedora and (White box Linux). I now have zap channels in one of the
boxes (T-1). No matter what type of channel I call on (sip or zap) I get
some really strange artifacts in the sound, almost like a skip in the
playback. It seems to always be in about the same place in the
recording. Usually in the beginning of playback. For
2007 Mar 14
3
What happend to voip-info?
Anyone has an idea what happend to voip-info? it stopped working about 24
hours ago.
Nir S
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2004 Apr 21
3
Webvmail
I am having trouble locating webvmail on my * server.
Is this a seprate porgram or does it come with *. I
am running version
asterick*CLI> show version
Asterisk CVS-03/26/04-17:08:20 built by
root@localhost.localdomain on a i686 running Linux
asterick*CLI>
Thanks
Kurt
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2007 Apr 25
3
FYI
Just been getting lots of failed SIP registrations to a system here.
All coming from Taiwan.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net
Euro Tech News Blog http://eurotechnews.blogspot.com
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 Mar 14
4
what happened to asterisk wiki???
Hi
im trying access the www.voip-info.org website since yesterday but i cant
open it. google search diaplay correct search results but it doesnt open
when i click the link. it displays a message about tcp error which says
-->"There was a problem communicating with the server". I dont know what the
problem is. I just want to ask whether their server is down or not and is
everybody
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 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
2007 May 31
9
click to call
I have been looking around for examples or code on making a click to call
application for web sites... has anybody had any luck on this topic? Is
there any open source code out ther that could do this?
Regards
AK
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 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>
2007 Apr 29
2
Early audio(progress) and MOH
Hi,
Is it possible to have MOH in early audio, while waiting for someone to pick
up a Dial() call?
(When using zap channels, I have early audio working with playback)
H?kon Nessj?en
Loopback Systems AS
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2004 Aug 17
4
Hunt Groups
I have a question about how Asterisk Parses the Dial Plan. To create a
hunt-group which would be the appropriate dial plan:
[CompanyABC]
exten => 7228888,1,Dial(SIP/8017228888,60,r)
exten => 7228888,102,Dial(SIP/8014361234,60,r)
exten => 7228888,103,Dial(SIP/8014362345,60,r)
exten => 7228888,104,Dial(SIP/8014363456,60,r)
exten => 7228888,105,Dial(SIP/8014364567,60,r)
exten
2007 May 28
2
Polycom Static IP
I am still having issues with my Polycom 301 phones when I disable DHCP. I
give the phone a static address and I keep getting the error 'could not
contact boot server using existing config'. As soon as I set it back to
DHCP enabled the phone can see the boot server and I'm online.
Steve
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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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2006 Feb 10
4
Sendmail with exchange
I am using Asterisk to send Voicemail out as Email. I am running into a
problem I believe to be caused by the exchange server requiring SMTP
authentication. I cannot get the sys admin's to turn it off. Does anyone
know enough about sendmail to help me. I am assuming that the default
mail client is sendmail. It will also send to other non-SMTP
authenticated servers. Your help is much
2007 Apr 26
1
Asterisk brute force watcher (was FYI)
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of J. Oquendo
> Sent: Thursday, April 26, 2007 6:47 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Asterisk brute force watcher (was FYI)
>
> Steve Totaro wrote:
> > I suspect that
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:
2007 Mar 08
3
Sender phone ringing while recipient talking
I've had asterisk running for about a month now between our PBX and our
T1, and everything seems fine but for one simple nit-pick: When a call
to the outside workd is made, and if the recipient picks up while a the
sender's phone is still relaying the ring, the sender won't be heard
until after the ring stops. This often translates a simple "hello?" into
a