similar to: driver model dude

hi list , is having problems when sending a fax with hylafax and a card sangoma A200D, when he sends it arrives to the destination but it paginates appears in white this is my log Aug 20 16:11:08 voz FaxSend[6715]: MODEM Supports 40 ms, 20 ms/scanline Aug 20 16:11:08 voz FaxSend[6715]: MODEM Supports 40 ms/scanline Aug 20 16:11:08 voz FaxSend[6715]: MODEM WWW.SOFT-SWITCH.ORG spandsp/ Aug 20

Hello, I am trying to add a sip-trunk to my Asterisk 1.4.15/Elastix 0.9.2 server. The system is in production with local extensions, a zap trunk and a working sip trunk with sipgate.de. My asterisk server is behind a NAT/Firewall, anyhow it registers and works well with sipgate.de on incoming and outgoing calls. I aquired an account with a reseller net-voz.com: I did some testing with the

http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I

Included 3 patches that updates documentation. This completes a 5 patches set. If you prefer me to resend all of then as one patch, attached, discussing or whatever, your're welcome. Best regards, vicente >From 7cec3ad78c8454408c8b6a1950d441e02d56d138 Mon Sep 17 00:00:00 2001 From: Vicente Jimenez Aguilar <googuy at gmail.com> Date: Fri, 23 Jan 2009 00:57:48 +0100 Subject: [PATCH]

Hello, in my project im using speex 1.2rc1 and the preprocessor VAD seems to only separate complete silence from not complete silence frames. The Speex Manual, you can read "The voice activity detector (VAD) provided by the preprocessor is more advanced than the one directly provided in the codec." but if you go to the source code in preprocess.c line 995 "/* FIXME: This VAD

For those who did not get the annoucement. CuteFTP is now claiming to support sftp as stated in the RFC. And at first glace it *SORTA* works.=) I can upload files, but the directory listings are broken (it does not display anything) and as a result download won't work (since cuteftp rquires the file to be in the directory listing to get). I'm going to back down a releas of

Jesus, Unfortunately, FFT and magic algorithms don't work (yet?). You might want to try this if you're not satisfied with Speex VAD: http://lists.xiph.org/pipermail/speex-dev/2008-August/006860.html It won't perform any miracles, but I think it works pretty well and is easy to tweak. Tom >---- Original Message ---- >From: jmorion at toomeeting.com >To: speex-dev at

The Asterisk Development Team announces the release of Asterisk 1.8.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release:

The Asterisk Development Team announces the release of Asterisk 1.8.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release:

Hi all. I've ordered a TDM400P with 4 FXO, but after using my X100P I'm thinking about returning the TDM400P because of bad echo issues. If I do get the echo issues I'll look at digital options. My question: Is anyone using ISDN (BRI) in the states? I've heard ISDN4LINUX devices suffer bad echo but chan_capi works great. All the chan_capi cards I find though are for overseas

-----Mensaje original----- De: Jorge García [mailto:jgf en sitrans.transnet.cu] Enviado el: Miércoles, 15 de Febrero de 2012 11:42 a.m. Para: 'Carlos Ortega' Asunto: RE: [R-es] Denominación alternativa para la que quiso llamarse "Comunidad de Usuarios de R": ¿ideas? Yo personalmente prefiero algo así como: HISPAN R IBER R ESTADISTICA+R ESTIA+R (Estadística Iberoamericana

Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco

The Asterisk Development Team has announced the release of Asterisk 1.8.4. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.4 resolves several issues reported by the community. Without your help this release would not have been possible. Thank you! Below is a sample of the issues resolved in this release: *

The Asterisk Development Team has announced the release of Asterisk 1.8.4. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.4 resolves several issues reported by the community. Without your help this release would not have been possible. Thank you! Below is a sample of the issues resolved in this release: *

Hi everybody, I'm trying to use msn with * and for that, I'm reading all information on the mailing list. You used to recommend the page http://mcleod.pbx.nq.net/msn/, but I always get an error while opening. Has it changed? Is there another one? Thanks cmayor ___________________________________________________ Yahoo! Messenger - Nueva versi?n GRATIS Super Webcam, voz, caritas animadas, y

Hi, Talking about messenger,,, it's still necesary to do HKEY_CURRENT_USER\Software\Microsoft\MessengerService\Corp2PC_Phone equals to '1' ??? But it's still sending the '+' digit, so it's necesary to stripMSD? Thanks a lot cmayor ___________________________________________________ Yahoo! Messenger - Nueva versi?n GRATIS Super Webcam, voz, caritas animadas, y m?s...

Hello I am making backup of a Plesk Debian server to /backup using Rsync. My questioin is how can I preserve the ctime, mtime, and atime of original files? Thanks

Hi, all My Asterisk uses a Digium TE120Pand I would like to add an Astribank zaptel_hardware sees is, but I cannot get it working pbx:~# zaptel_hardware Argument "IRQ" isn't numeric in numeric comparison (<=>) at /usr/local/share/perl/5.8.8/Zaptel/Span.pm line 114. usb:005/002 xpp_usb- e4e4:1131 Astribank-8/16 USB-firmware pci:0000:04:00.0 wcte12xp+

Anyway to make hitting `0` during a voice mail dial an extension? The bosses used to have that feature and love it. Their VM prompt would say: "Hello, My name is blah blah. I am currently unavailable. If you would like to speak to an operator press 0 now, otherwise leave me a message". Extension 0 exists, but dialing it during a VM prompt does nothing. Thanks, -- Daniel Jimenez

Hello list, I think I have a rails related RSpec problem with a mock going out of scope on a recursive call to a model. The code is at: http://pastie.textmate.org/79821 if you want to see it highlighted. I have pasted it below as well. Basically, I have an acts_as_nested_set model called "Node", which works fine. I have a function which finds the language name of the node instance.