search for: polybeacon

Displaying 20 results from an estimated 175 matches for "polybeacon".

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. -------------- next part -------------- An HTML attachment was scrubbed...
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 Jul 09
2
Re : Re : Re : Communication IAX2 >SIP>IAX2
...omain=domaineName [100] secret=100 username=100 type=friend context=pstn2 nat=yes disallow=all allow=ulaw allow=gsm allow=alaw host=dynamic i do not know why it prints No such host: pstn2/011212664800450?? Any suggestion ________________________________ De : Paul Belanger <paul.belanger at polybeacon.com> ? : Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Envoy? le : Jeu 8 juillet 2010, 12h 10min 14s Objet : Re: [asterisk-users] Re : Communication IAX2 >SIP>IAX2 On Thu, Jul 8, 2010 at 6:29 AM, Adil Zaaraoui <adilzeaaraoui at yaho...
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100914/b56c3d9c/attachment.htm
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 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? -------------- next part -------------- An HTML attachment was
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 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100821/4d11d6c0/attachment.htm
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* --------------
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 08
3
IPSec on asterisk
Hi, I am trying to configure ipsec on asterisk. Have configured /etc/racoon/racoon.conf and /etc/raccoon/psk.txt. Also have policy file in same folder. Have run racoon. Still I can't receive calls. Can anyone please tell if any extra step is needed. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: