Displaying 20 results from an estimated 10000 matches similar to: "Sender phone ringing while recipient talking"
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 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 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 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 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
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070314/23cdc0f6/attachment.htm
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.
-------------- next
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 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070429/c901ff90/attachment.htm
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 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 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
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.
2007 Apr 24
6
Digium card sale
Good morning,
Pardon for this intrusion I just wanted to let everyone know about some of
the specials that I have going on at HYPERLINK
"http://www.astawerks.com"www.astawerks.com . From now until the end of
June I will have a huge unpublished sale on all Digium products. Prices are
way to low to list so I will have to be personally contacted. I also have a
permanent sale on all
2007 May 30
1
Application Developer
I'm looking to hear from any application developers in Argentina
specifically or other South/Central American countries.
Please understand this isn't a dial plan or remote installation I'm
looking for but an actual application developer.
If this fits your description please email me details on;
Size of company (number of full time/versus contract staff)
Location
Previous
2007 Mar 23
7
Doorphone vs. Grandstream BT101
I've done all the googling I can on this, and have come to the
conclusion that a Grandstream BT101 can be abused to be a door phone.
Could someone with access to one, confirm that the following is possible?
Researched:
1. When set to auto-answer, dialing the phone will result in a short
beep and instant speaker-phone connection.
2. When pressing the "message" button while
2007 May 08
2
asterisk with festival facing problem
hi List,
i've been trying to get festival work on my 1.4.4 *box for the last 3days,
i've used the tutorial on this page
http://www.voip-info.org/wiki-Asterisk+Festival+installation
with exactly the same line in my dialplan just to make a test
now when i try to call( dial 555 ) from my softphone i get this message on
festival server debugger:
server Tue May 8 11:36:53 2007 : Festival