Displaying 20 results from an estimated 3000 matches similar to: "ASterisk CDR file Master.csv"
2010 Oct 22
5
dials a trunk when off hook
How can I let asterisk immediately dials a trunk when off hook?
2012 Sep 26
1
Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH
Hello,
I'm having issues connecting throu PRI with the following error "Requested
transfer capability: 0x00 - SPEECH"
Below are the logs:
== Using SIP RTP CoS mark 5
-- Executing [97052660 at voipphones:1] Set("SIP/4856-00000003",
"CALLERID(num)=xxxxxxxxx") in new stack
-- Executing [97052660 at voipphones:2] Dial("SIP/4856-00000003",
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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2011 Jan 24
1
B410P: DAHDI BRI PTMP HDLC Abort (6) on Primary D-channel of span 1 (TEI Errors)
Hi all,
So, we reverted the LibPRI version and tested it, and then tried with
the latest version of everything. Still no changes.
The BRI line is in PTMP. If we set the configs to PTMP in the
genconf_parameters and try it, we get the following:
[Jan 21 17:32:20] ERROR[20341]: chan_dahdi.c:12645 dahdi_pri_error:
Unable to receive TEI from network!
If we set it to PTP (which it is not) we
2010 Oct 16
3
Detect incoming fax on PSTN and route to fax machine on DADHI extension?
I'm running an AsteriskNow V1.7.1 with both a PSTN connection and fax
machine. Both are connected to a DAHDI board. I'd like to route
incoming PSTN fax calls to the extension of the fax machine and process
non-fax calls through different dialplan.logic.
What's the best way to go about doing this? I've looked into Fax for
Asterisk, bit I'm not sure that I want it or NVFax
2005 Oct 10
2
Asterisk and Mitel SX 200 Slip and Frame Errors causing Major Ala rms
We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get
over 500 frame errors and over a 500 slip errors per hour. When the errors
reach 1000 per hour the Mitel will take it's T1 card offline. At that point
no calls can be routed from the Asterisk server to the Mitel and the TE110P
reports a Yellow alarm.
What can be causing all these Frame and Slip errors? We have been
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 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 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 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 Aug 06
1
Reinstalling Asterisk due to hardware changes
My purely SIP experiment has failed so I am purchasing a Digium E1/T1 card to put into my Asterisk box.
I know from the wonderful O'Reilly book that the proper installation is Zaptel ? libpri ? Asterisk. Is it possible to simply reinstall in that order once I have installed the card and have Asterisk successfully work or is it best to uninstall all three packages and install from scratch?
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
2010 Oct 15
2
Echo on PRI after upgrading to Asterisk 1.6.2 from 1.2
I've recently upgraded an Asterisk system from 1.2 to 1.6.2 (did a
full reformat and recompile) and I started getting echo over the PRI.
I've tried the default settings for echo in the system.conf file as
well as I've compiled OSLEC to try and see if thats any better.
I'm not sure what to try next. Does anyone have any suggestions?
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 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 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 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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