Displaying 20 results from an estimated 4000 matches similar to: "FYI: Seen the 2600Hz announcement?"
2010 Aug 24
4
1.6 and asterisk gui
Hello,
I'm new to asterisk and this list. The ISO download appears to have
1.6 with the FreePBX GUI but I am looking to use the Asterisk GUI.
The only option for the Asterisk GUI is to use 1.4. Is it as simple
as installing 1.6 only then using the yum repository to install the
Asterisk GUI? If so, what packages are needed?
Thanks!
2010 Sep 16
4
[OT-FreePBX] Outbound calls check inbound routes to see if destination is local?
Greetings-
First, my apologies for the OT post. Yes, I understand this is not the FreePBX-users mailing list. But, there are a large number of people that use FreePBX and I'm hoping they can be of assistance.
I have a system running Asterisk 1.4.27 (see... relevance!!!) and FreePBX 2.6.0. There are a large number of inbound routes configured for the various DID's coming in via PRI, SIP,
2010 Sep 11
8
No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Hi Everyone,
I have a provider whose DID used to come into the box just fine but recently
stopped working. Nothing has been changed on our end.
Here is what I get when doing "sip set debug peer PROVIDER":
Sending to 123.123.123.123 : 5060 (no NAT)
^^^^ That is ALL I am getting with sip debug turned on.
With Allow Anonymous SIP set to YES, then the call comes in properly and you
see
2007 Mar 16
1
FW: Microsoft buys Tellme
http://deancollinsblog.blogspot.com/2007/03/microsoft-buys-tellme.html
I thought I would email this post I made on my blog from yesterday as a
way of stimulating discussion on this.
It looks like the Asterisk community is no closer to getting a Pre-Paid
'Offboard Speech Recognition Processing' SIP gateway than ever. I've
left the
2010 Sep 09
5
info about application not available asterisk 1.6.2.11
Hello list,
how come on my Asterisk 1.6.2.11, I have no help available ?!
asterisk*CLI> core show application Dial
-= Info about application 'Dial' =-
[Synopsis]
Not available
[Description]
Not available
[Syntax]
Not available
[Arguments]
Not available
[See Also]
Not available
Kind regards,
Jonas.
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2010 Oct 21
10
Asterisk 1.80-rc5
Just done a clean install of rc5 on a totally new machine and found the
following:-
/etc/init.d/asterisk start
errors on line 109 - there is no 0 before $VERBOSITY as in the other lines.
More interesting is that after make samples I have no iax2 available.
Dave Cotton
2010 Jul 24
4
getting some segmentation faults with 1.8
I downloaded the latest 1.8 (27922) but got some segmentation faults.
The first one was when it loaded cdr_odb, and so I changed menuselect
not to compile that one, but the second one was when it tried to load
chan_agent and so I stopped there to see if anyone else was seeing
this. The agents.conf is all commented out except for [general] .
Anyone know what is happening?
Thanks.
P.S. I deleted
2010 Oct 14
6
Audiocodes firmware
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body text="#000000" bgcolor="#ffffff">
<font size="+1">Does anyone have links to the most recent audiocodes
2010 Oct 22
5
dials a trunk when off hook
How can I let asterisk immediately dials a trunk when off hook?
2010 Oct 18
5
Same extension registering over eth0 and eth1
Hello list,
I need to know how to deal with a redundant network with only one asterisk
server, which is receiving registrations from the end points on both of its
ethernet ports. This means extension 201 is registering both from eth0 and
from eth1.
Is there a way/software which can act as a middle man between asterisk and
the ethernet ports, and by default sends registrations to asterisk only
2010 Sep 16
4
one way audio for xlite clients behind NAT
I am having a one way audio issue with xlite clients behind NAT. They can
connect to the server and make calls but no audio is heard on the other
end.
my sip conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" <XXXX>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
2010 Sep 14
9
Speech To Text on linux with asterisk
Hi,
Is it possible to record say 30 seconds of audio and then have LumenVox
convert to text ?
or any available tool open source for speech to text .
Regards
Dhaval
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2010 Oct 24
5
Integrating Asterisk 1.8 with Google Talk and Google Voice
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks
2010 Jun 26
2
Detecting hook flash in asterisk
Hello,
Is it possible to detect a hook flash in asterisk. I want to be able to
perform some functions an hook flash.
I have the following entry in features.conf which executes a Macro on
detecting key press '**'.
[applicationmap]
test => **,caller,Macro,testflash
Is it possible to do this action on hook flash?
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2010 Oct 23
2
Just Take dCAP at Astricon?
Since it is Saturday evening (7PM EST) I am asking this on the list in case
someone who knows sees it and can answer.
Astricon is in my back yard for the first time, and I could hit you with a
rock. I would always like to attend, and spoke at the 2007 Astricon in
Phoenix but don't have the idle cycles.
Question: Can I just go to Astricon and take the dCAP exam only? In and
out? Cost?
I
2010 Jul 23
6
Asterisk 1.8.0-beta1 is Now Available!
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1.
This release marks the beginning of the testing process for the eventual release
of Asterisk 1.8.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any
2010 Jul 23
6
Asterisk 1.8.0-beta1 is Now Available!
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1.
This release marks the beginning of the testing process for the eventual release
of Asterisk 1.8.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any
2010 Aug 21
7
Opensource Speech recognition for Asterisk
Hi Everyone,
Has anyone got any opensource speech recognition software to work with
Asterisk? Please only list WORKING ones. Not the "theoretically" should work
ones!
Thanks
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2010 Jul 09
2
Re : Re : Re : Communication IAX2 >SIP>IAX2
ok it works i had a problem with a syntax:
i had to wrire:
exten =>_!X.,n(external),Dial(SIP/011212664800450 at pstn2,,S(20))
thanks
________________________________
De : Adil Zaaraoui <adilzeaaraoui at yahoo.fr>
? : Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Envoy? le : Jeu 8 juillet 2010, 19h 41min 15s
Objet : Re :
2010 Sep 08
2
Max TDM calls per asterisk box
Hi Everyone,
Can you tell me how many concurrent TDM (Dahdi) calls
that a single asterisk box can handle.
Configuration is as follow :
Quad core Xeon 3 GHZ, 4Gb RAM, asterisk 1.6.2.9
Also do you know a good tool to stress out asterisk?
Kind regards
--
*Adolphe CHER-AIME
Network / VoIP Engineer
CCNA, CCNA VOICE, Global VSAT Forum Certified
(509) 3449-4280*
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