Displaying 20 results from an estimated 7000 matches similar to: "Curious what 'early media' is in terms of Answer()"
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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2011 Jun 24
3
t.38 virtual fax software?
Can anyone recommend some kind of virtual t.38 fax software? I'd like
to test/debug some of the t.38 stuff, but it'd be much easier if I had a
software client that could just generate the faxes from a workstation,
rather than having to sit with the fax machine + t.38 ata to source
faxes from.
There doesn't seem to be much out there, and the stuff that's out there
is kind of
2010 Aug 23
2
DAHDI not detecting caller hangup
Hi,
Odd problem have just noticed in that when I call into the PBX DAHDI detects the call and hands it off to the extension, if I then hang up it still continues to process through the dialplan.
This is what I have in chan_dahdi.conf:
[channels]
language=en
echocancel=yes
usecallerid=yes
cidsignalling=v23
sendcalleridafter = 2
hanguponpolarityswitch=yes
rxgain=2.0
txgain=3.0
progzone=uk
2010 Sep 10
7
A way to check against a list of numbers?
Does anyone have a suggestion on how to handle this? For example, if I
have a list of numbers that I want to go out a certain sip channel and
another that I want to go out the dahdi device, is there a way to do
this? None of the numbers will fit into a pattern, so just plain
pattern matching won't do.
The most straightforward way would be to just define explicit patterns.
Obviously that
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 Sep 12
1
First boot asterisk -vvvvvgcn segfaults
Hello all,
I have recently finished compiling and making asterisk on a fresh
slackware 13 machine. I went through and made sure to install ncurses,
zlib, openssl, libnewt, libpri, and dahdi prior to compiling asterisk.
I configured with this:
$ ./configure --prefix=/usr --enable-threads=posix
--target=i486-slackware-linux --host=i486-slackware-linux
Now that I thought Asterisk was ready to go,
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
2014 Jan 31
1
Asterisk as a media gateway
I'm playing around in a lab, and I was wondering if its possible to have Asterisk act similar to that of a Avaya PBX, where we have media gateways do the heavy lifting.
This is what I was thinking of trying.
1. One asterisk server will contain the logic of the phone system (ex: queues, extensions...etc).
2. The mains server will not handle RTP traffic, it will send the RTP traffic to
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 Sep 14
6
Spontaneous reboots on asterisk 1.6.2.11
Hello list,
has anyone else also noticed spontaneous reboots ?!
I noticed this today and also yesterday. Can't really see if there is a
fixed time between the reboots.
Normally al of my SIP peers are registered. When I put up the CLI today
I saw that a lot of SIP accounts where UNREACHABLE and needed to
register again (what they slowly did).
These are realtime SIP peers that reside on
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 Oct 22
5
dials a trunk when off hook
How can I let asterisk immediately dials a trunk when off hook?
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 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 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
2011 Mar 31
1
Transfer feature dialing out after one digit
Because some users have requested transfer beep confirmations I've
switched our phones over to using the asterisk transfer feature instead
of the built in transfer functions of the phones. While testing it was
working fine, but I changed something in features.conf and suddenly any
time I hit transfer (*2), I can only enter one digit before asterisk
immediately tries to dial that extension.
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