Displaying 20 results from an estimated 2000 matches similar to: "what happened to asterisk wiki???"
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
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
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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 30
12
False ring problem
Hi all,
when a user dials any number, asterisk automatically generates ringing which
caller can hear, and after 2 - 3 rings asterisk detects that the called user
is busy, then caller hears busy tone. for example user hears---
tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing
at the start so that user hears only beep beep beep if the called user is
busy. I have used the R
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
2008 Aug 06
2
shared mysql connection in dialplan
hi all,
i just finished developing some incoming call features in a macro. that
macro gets executed everytime an incoming call is received and a new mysql
connection is made using the MYSQL cmd in dialplan. i want to use a single
mysql connection for every incoming call.
my idea of doing it is like this, i want to get a mysql connection in a
global variable, just to share the connection with
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 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
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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 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 Apr 19
1
CDR(dst) != CALLERID(dnid)
Hi guys,
i just came to know that CDR(dst) field is set to current extension instead
of the dialed no. i need to set it to DNID because our every user has 5 dids
and i want to show the caller at the end of the month which numbers he
dialed for every call, along with other cdr info. Our rating depends on the
dialed number also. here is my extensions.conf
exten=> 1212,1,Dial(SIP/rizwan)
2007 Aug 17
4
Call Limits
Hi all,
Some of my asterisk users have used their maximum call limit for incoming
calls (peers). There incoming call limit should automatically reset to zero
after hangup but its not happening and they no longer can recieve any calls
as their allowed limit is already full. So is there any way to reset the
call limit on peers by commands or do i have to restart my asterisk server?
--
Best Regards
2007 Sep 11
3
Prevent multiple sip registrations
Hi all,
Is there anyway i can prevent multiple sip registrations from different IPs
using single username in asterisk. Does asterisk provide any aid in this
respect? As far as my knowledge is concerned i dont think there is any
support for this in asterisk, so i think i'll have to makeup a script which
sniffs sip packets coming for asterisk and detect for multiple register
requests coming from
2007 Oct 24
2
Remote provisioning for ATA's
Hi all,
I need a fully developed web based remote provisioning system. I cant find
anything reliable on the internet. Have already checked ataconfig.com and
voxilla-ays.com. have tried to contact them but got no response. So if
anybody knows a good provisioning system then plz tell me about it.
--
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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