similar to: Replacement for Allison

Hi folks, With thanks to Alison Keenan (another Alison!) for the voice, Chris Bagnal for converting from 44k wav to sln and finally Terje Elde for debugging my HTML code, the British English files are now ready for download. They can be got from http://www.enicomms.com/cutglassivr/ Thanks and don't forget to practice safe IAX ;-} Mark -- Mark Phillips <g7ltt@g7ltt.com>

Hi Does anyone know if there are any free UK accented English sounds packs? Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Hi folks, I've posted uLaw, aLaw, G729 and G723 variants of the Alison Keenan British English files. http://www.enicomms.com/cutglassivr/ Thanks -- Mark Phillips <g7ltt@g7ltt.com>

Hi All, I'm going to be upgrading our 1.2 Asterisk system. At the moment we use the Enicomms SLN files. Are there major differences in the 1.4 default voicefile packs, or will I be able to re-use Enicomms?? In the Make menuselect, I noticed theres no .SLN voicefile selection for the basic audiofiles - has SLN been depreciated? Thanks Adrian

Mark, While these samples are pretty good they do not work "out of the box" - there are a couple of issues: 1. the samples are 44100 samples/second and Asterisk needs them to be at 8000 samples/second. This is what happens if you prune out all of the Amercian voicemail prompts and substitute yours: Asterisk 1.2.7, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark

Hi, I was running asterisk 1.0.7 but we've upgraded now to CVS-HEAD. I've noticed this.. and several people have commented that audio quality seems to have gone down hill. Just going phone-->asterisk-->PRI. I've not changed the configuration files during the upgrade. sip.conf is: allow=ulaw allow=ilbc allow=g726 allow=g729 allow=g723.1 And all the phones had been using

Hi, I have a couple of samples that produce interesting artifacts when encoded with the CVS snapshot of 2001-02-13. Both are about a meg. ftp://slumber.dhs.org/tmp/4.wav.bz2 When encoded with oggenc -b 128, there is a sort of stereo separation in the drums, while in the original they are 'solidly' positioned in the stereo field. This also occurs with -b 160 and is barely audible with

Hi,I have been using wav files with sample rate of 8khz and 8 bits and I find the sound quality really poor.What is the best sound quality I can achieve on Asterisk?Responses would be appreciated.Rgds,Akpome _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE!

Can anyone recommend a company that does professional Asterisk recordings for things like IVR, greetings, MOH, announcements, etc? Thanks, Waldo

Does anyone have a copy of spc.exe they could send me? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061117/ac0b6a44/attachment.htm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Here are the details: Successful call: INVITE cseq 1 From NexTone 100 Trying cseq 1 From Asterisk 100 Trying cseq 1 From Asterisk 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone INVITE (G711U)

I have a problem where SIP calls from some providers are dropping at 15 minutes. The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server. Below, 'Client' is the IP address of the client's host (running FPBX-2.8.1(1.8.20.0) 'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls 'Asterisk' is the IP

Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX -

Hello, This is not exactly an Asterisk question, but I was encouraged to seek advice here anyway. The kindness of the * open source community is legendary :) I am getting going with an Asterisk 1.2 box, and I'm having trouble getting good quality transmit sound using handsets with VoIP phones. I'm primarily trying to focus on SPA-941, but also experimenting with Aastra 9113i and Uniden

I just installed Postfix, MySQL, and Dovecot. Everything is working great (IMAP, POP, SMTP) except that I can't get incoming mail to deliver properly. Incoming mail logs the following: Feb 2 13:23:52 mail postfix/qmgr[2187]: CE0D41F0263: from=<mhoppes@[redacted]>, size=650, nrcpt=1 (queue active) Feb 2 13:23:52 mail postfix/pipe[3594]: CE0D41F0263: to=<mhoppes@[redacted]>,

I installed G729 from Diguim and I was expecting the sound quality on my i686 machine to be better than gsm. Compared to gsm, G729 sounds closer and a little robotic. Is this what is supposed to be or am I missing something? I am interested in G729 because the internet in my country is very expensive and I want to save every bit possible. I want to use G729 because it takes less bandwidth for

I'm looking for a wireless desktop VoIP phone. Does any exist?

Hi, Did anyone hear about a recent ruling which makes it illegal to have caller-id set to anything except what is on the account of the user? IE... If your name is "Joe Smith" you can't have "Mary Smith" set as the caller-id name, unless mary smith is also on your account.

I am running * 1.0.5 and have been having lots of problems with outgoing calls and their sound quality. I am using ULAW for the codec and sixtel for termination. Basically the problem is that portions of the call seem to be lost and replaced with silence. Sometimes I can't hear the person talking othertimes they can't hear me. This situation comes and goes throughout the call. Bandwidth

Anyone have an example XML file for the PAP2T? Cheers, j