search for: kenyon

...a a Cable Modem. When I try to start SWAT netscape displays the message that it cannot find the local host on port 90. I have downloaded the book over samba and I have also tried to edit the swat config file. I would really appreciate any and all information on what could be the problem. Thanks, Kenyon Wessman email: kwessman@woh.rr.com

I need to set up the sound card of a server to use in an overhead paging system, as normal I am testing this on my home machine first (which has slightly different Hardware). I'm using chan_alsa with the Intel HD Audio driver on an Intel 82801G (ICH7 Family) sound card. I am running Asterisk 1.4.17 and have a fully loaded TDM400P as a timing source. When calling console/dsp (using

Hi, I am a beginner who began to use Asterisk in this July. Last week, I made two addons for PostgreSQL (cdr_addon_postgresql.c and res_config_postgresql.c), because I use not usual MySQL but PostgreSQL. # Of course, not scratch build but modified version 1.4.8. But I don't know how to change configure script, menuselect-xxx, ... etc, and how to merge my sources to addons tar-balls. Anyone

Is there an advantage to having a Queue members URI in the form: SIP/User (or indeed IAX2/User) Over Local/<number>@context ? I know that the latter will allow you to do things like set counting logic etc. through dialplan operations, but the former appears to be a more direct route to calling the party. (and if need be, there is the ability in queues to run a script on connection iirc).

...for time. I have a working proof of concept program written in Java, using JMF and a port of the Vorbis decoder to the JMF (J-Ogg). Is there a bit pattern that would represent a silent Vorbis packet for any arbitrary stream? The null packet (ie all zeros) also has 0 length. Thanks, -- Daniel Kenyon Jones Emmanuel College, Cambridge dk282@cam.ac.uk --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-dev-request@xiph.org' containing only the word 'unsubscribe' in th...

I have a TDM400 card with 3x FXO and 1x FXS ports on it. At the moment I'd prefer (till I can get it working more reliable with iaxmodem), for a faxmodem to answer one of the lines instead of the linecard. I've tried changing the context of that line so that the exten => s does nothing, but that stops the line from being able to receive calls (get a recorded This number is not

There is a storage issue with the hoster. Things should get back to normal soon... hopefully! Arno (sent from my S3... please excuse my brevity) Le 8 oct. 2014 00:22, "Paul Kenyon" <pkenyon at advancedclustering.com> a ?crit : > Hello, > > I see you're a member of networkupstools on github. Do you have any > contact with the folks who manage the website? Getting a 503 error for all > pages. > > -Paul > > -- > Paul Kenyon > A...

I know I must be being daft, but is there a way to set which context the queuing system uses when it dials the operators/agents? By default it appears to use the default context. I've looked through voip-info.org and can't find anything, someone please put me out of my misery.

Is there any news on getting the Spam Filter fixed for this mailing list?

Hi I have updated a 10.6.8 to 10.8.4 and it transferred the dovecot mail. All except one sent folder. This folder was a sent mail box and the CUR folder is in the correct place and i have deleted the index files but it still is not showing up in a mail app. Can someone suggest a way to re catalogue it. many thanks jonny

I have attached a screen shot of the error I am running into. I am on page 138 of the Agile Web Development with Rails book by Pragmatic Programmers. I am developing the depot store, and I ran into an error, with hidden_div_if. When I remove that block of code, things to work fine, besides the fact the cart does not display on the site, but everything else functions correctly, so it is

...e comment on that? If the source code says "deprecated" but the CLI help does not mention that - whom do I trust? -------- Original message -------- Subject: Re: [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions From: Philipp Kempgen <philipp.kempgen@amooma.de> Thomas Kenyon wrote: > Philipp Kempgen wrote: >> You might use the Blacklist() application in 1.2 (deprecated!). >> Using AstDB is an option: >> > > Is it really deprecated? I use LookupBlacklist (tested as working) in > 1.4, and show application LookupBlacklist doesn't mentio...

I am having a problem trying to get a particular softphone working on my setup. The machine it runs on has more than one interface. When the softphone registers, it registers fine, and asterisk is given the correct IP for registration. Whenever RTP is set-up however, the client gives the wrong IP to connect to and I get the inevitable problem with one-way media. Is there any way of forcing

I'm about to upgrade my currently running asterisk server from 1.0.3 to 1.0.6 is there anything I should do before doing the upgrade? I know since I'm using Fedora Core 3 that I must do for Zaptel : make clean then make linux26 and finally make install. Do I have to remove the version 1.0.3 first or only doing install will replace all existing modules without removing my config files?

I need to be able to connect an old PA system to an asterisk box, which basically works as a couple of amplifiers taking an analogue phone signal and playing whatever it produces out of some speakers. There is no on-hook state in the whole setup. Obviously If I just connect the input to a port on an ATA, I'll just get a dialtone played through the speakers. Can anyone think of a way I can

I know many of you have been waiting for this for a while, so I'll keep this short: The Skype for Asterisk Public Beta is now available on the Digium store. We are pleased to announce the open beta of Skype For Asterisk is ready to begin and we look forward to you participation. To obtain your copy of the software, please visit Digium?s web store and purchase (for zero dollars) the

Asterisk 1.2.24 seems to crash repeatedly under any substantial call load (and sometimes without a substantial call load - just one SIP leg is enough to do it) when using the G.729 pre-compiled binaries from: http://asterisk.hosting.lv/ As per: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing Time to crash is variable, but seems to require at least an hour of production performance

This may sound silly and easy, and in fact I hope it is, but I'm having trouble setting up a web link to activate a stream. I have tried using formats I knew of, like the format used by Shoutcast (hots:port/shoutcast-playlist.pls? and host:port/listen?pls) but have had not luck. I was wondering if anyone out there knew how to do this, and I would really appreciate any help. Thanks. Rob

...n a few years. My drive is filling up and all, I probably have tons of old/vulnerable daemons, so it's probably time for a fresh install. I'll let you know what turns up! -P -----Original Message----- From: Jack Moffitt [mailto:jack@xiph.org] Sent: Wednesday, August 08, 2001 1:29 PM To: Kenyon, Paul Cc: icecast-dev@xiph.org Subject: Re: [icecast-dev] IceCast 1.3.11: Segmentation Violation in Linux > Linux poseidon 2.2.15 #1 SMP Wed Mar 21 16:48:40 CST 2001 i686 unknown > libc version is 5.4.46 Not to be a bummer and blow you off, but we won't be supporting libc5. It's too...

I have an asterisk 1.2.9.1 machine with zaptel 1.2.6 running. On the TDM400P, I have 1 FXS port and 3 FXO ports. dmesg reveals: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.6 Echo Canceller: KB1 PCI: Found IRQ 10 for device 01:01.0 PCI: Sharing IRQ 10 with 01:05.0 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: