Displaying 20 results from an estimated 122 matches for "asteriskhelpdesk".
2007 Mar 07
4
OT Vonage V-Phone Adapter (Possible Hack)
...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
/rpsm/oid/162059/rpem/ccd/productDetailAccessory.do#tabs
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
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
...be
missing something on where to enable it, right? The readme says
nothing.
This box is fedora core 6 with all the bluez stuff installed and loaded
and a dongle attached. I can see and pair with the box with my cell
phone so Bluetooth is working in linux.
Ideas?
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
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
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
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 Jun 06
3
1.4 Zaptel/Sangoma Issues on CentOS
...t) ]
> 0 bytes of data
-- Restarting T203 counter
q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:664 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED
-- Restarting T203 counter
== Primary D-Channel on span 1 up
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
2007 Mar 08
0
Fwd: Back to back E1 - asterisk <=> toshiba pbx -Call droping
...nected directly to
the telco you will want to use '1' to accept timing from them. If
youhave multiple spans, set them as 2, 3, 4, etc.
Problems with timing manifest themselves different ways - with static,
pops, and channels or calls regularly dropping.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
_____
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Vidura
Senadeera
Sent: Thursday, March 08, 2007 1:01 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Fwd: Back to back E1 - asterisk <=> toshi...
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
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 Apr 26
1
Asterisk brute force watcher (was FYI)
...hat
many
> > people who have weak SIP credentials like user=100 secret=100 will
be
> > the victim of toll fraud and worse, call to 900 and other very high
> > termination rates. How does $25 per minute sound?
> >
> > Thanks,
> > Steve Totaro
> > http://www.asteriskhelpdesk.com
> > KB3OPB
>
> Ashtray is an Asterisk brute force watcher. Checks logs from cron and
> emails admin of potential brute forcers
> http://www.infiltrated.net/scripts/ashtray
>
> Can have it set in .bash_profile so whenever you log on, you'd see
> anomalies.
>...
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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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 May 19
2
(OT) Anyone Ever Use http://shopfort1.com as a Broker
...thing good or bad to say about this outfit. Quoted
prices are really good for my needs. Much cheaper than the broker I
have always used in the past.
They will not tell you the carrier until you speak with a sales person
but "It's all Verizon (tm)".
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
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 Mar 14
2
Manager connection problems
I am wondering how many and how often manager connections can be setup
and torn down reasonably.
here is the scenerio...
I have 10 to 20 agents on two queues
one with priority over the other
I changed this the day before
I also implemented a php program that runs every 8 seconds on an
automatic refresh
It establishes a connection to asterisk and runs a mysql query to update
the database
2006 Mar 27
0
Question about Polycom 601 and expansion module.
...achment was scrubbed...
> URL:
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> /attachment-0001.htm
>
> ------------------------------
>
> Message: 2
> Date: Mon, 27 Mar 2006 14:20:49 -0400
> From: "Steve Totaro" <stotaro@asteriskhelpdesk.com>
> Subject: [Asterisk-Users] Ability to put call on hold via manager?
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users@lists.digium.com>
> Message-ID:
> <DFB93BD730105941BD1A782A1EE9E95CCC07@1-0fa9e300af524.asteriskhelpdesk.c...
2007 Apr 19
5
Polycom IP 501 is displaying wrong time
Hi,
This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the display screen. How can I set the "New York" time? What value I have to give to GMT offset value?
Look forward to your response. Thank you.
Regards,
Chandra.
---------------------------------
Ahhh...imagining that irresistible "new car" smell?
Check outnew cars at Yahoo! Autos.
2006 Jan 25
0
ISDN / Analog
...since doing this
over VoIP/Asterisk can be problematic. It is best to keep these lines
out of the PBX.
I have done several implementations on IBM x305 and x306 servers and
they work great. I cannot comment on the x100 though, since I have no
experience with it.
Thanks,
Steve Totaro
www.asteriskhelpdesk.com <http://www.asteriskhelpdesk.com/>
_____
From: phil.dawson@marnock.com [mailto:phil.dawson@marnock.com]
Sent: Wednesday, January 25, 2006 6:34 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] ISDN / Analog
Hi,
We have 12 lines via ISDN30 but this seems...
2006 Mar 26
2
Web based voicemail client
I'm looking for a good web based voicemail client that can use mysql or
realtime drivers. I can't seem to get vmail.cgi to work with realtime.
Thanks for any help you can give.