similar to: British English voice files are ready for download

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>

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 Folks, I have a few folks whom are interested in another recording session with Alison Keenan but don't have enough work to justify her visit to the studio. If there's anyone whom would like her to do some work but hasn't got around to it yet now might be the time. We need enough work to fill an hour of her time. So far we have about 25 minutes. Le me know off list. Thanks --

Hi folks, I have British comedienne, Alison Keenan (another Alison!) coming in on Saturday afternoon to record the Asterisk prompts for me. Alison speaks with a "posh boarding school" accent. Finally we'll have a free British English female voice bank. As I have her in my studio (yeah right; it's a cupboard under the stairs) does anyone need anything doing? She's charging a

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, Does anyone know what I need to do to get these: http://www.enicomms.com/cutglassivr/ Sounds files to work? I've tried loading them, but they are completely silent (format mis-match maybe?). Specifically, when I try to enter voicemail, nothing plays... though it clearly tries. I'm looking for replacement sound files for the default Allison, as I feel she is kind of breathy. I have

I run an Apple OS X workstation and I've got a server on the same LAN that's both a webserver and an Asterisk PBX. I wanted to be able to originate calls in the OS X Address Book application, and have Asterisk dial them and connect them to the phone on my desk. I've assembled a system that uses AppleScript to connect, via XML-RPC, to a web application that, in turn, connects to

Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

I just got a flyer from my buddy on these phones today, totally SIP based, includes the G.729 speech compression codec. http://dm.zipphones.com/dm/zip2/index.htm Any word on these? -- Mark Street, D.C. Red Hat Certified Engineer Cert# 807302251406074 -- Key fingerprint = 3949 39E4 6317 7C3C 023E 2B1F 6FB3 06E7 D109 56C0 GPG key http://www.streetchiro.com/pubkey.asc

Does anyone know how many simultaneous calls can a WRTG54GS handle? Assuming SIP phones are connected locally using G711.u codec and the WRTG54GS connects to a remote Asterisk server using IAX2 trunking using GSM codec. Thanks, Daniel

I have purchased a single G729 license - however, how many are actually needed? All my IP phones have G729a codecs built in (Cisco 7960 / Zultys ZIP2) - I would have assumed that if the phones can do it, and canreinvite=yes, then the phones shouldn't need to go through asterisk anyway? For calls that do go through asterisk, is a single license required for each side of the stream? (i.e. a

Hi all, Does anyone know of a method of sending a raw G711 stream to an address in Asterisk. For example, an application that takes a argument of a phone and a port. The reason? I have found a method to paging on Zultys ZIP2 and ZIP4x4 handsets. Basically it involves sending a stream of RTP data to port 3771 to multicast address 224.0.0.1. Would it need to involve me writing my

I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't

All, We have a fairly vanilla Samba configuration that recently replaced a Windows 2003 server and among other things, serves large (>64 MB) files. Permissions are all 777. When running an application attempting to do a single read of these files from a share, we discovered that they were not being served properly. We also found that copying them to the local drive or changing the ownership

I tried to encode some video from a DVD with encoder_example and ffmpeg2theora, and it looked great, except that with both encoders the A/V sync would drift noticeably after about 10 minutes, and by the end of a two hour movie it would be a couple seconds off. So instead of just whining about this and waiting for someone else to fix it, I decided to try to learn about what's causing the

Hello, I have a bunch of Polycom 601's and Asterisk 1.4.13. The problem is that the MWI indicators will never go off (The blinking red light and envelope in the LCD). I have tried to upgrade to 1.4.14 and all different SIP versions on the Polycoms. I am now at 1.6.7 Here is the SIP Message that turns on the lights: Scheduling destruction of SIP dialog '

Hi -- I'm working with a number of finance companies in New York City and Chicago which are looking to develop languages and platforms in order to process extremely large datasets. As such, we are looking for people with experience and / or interest in programming language and compiler design -- hence my posting on the LLVM mailing list. Goldman Sachs most famously developed Slang and

Hi, When I write various test cases and explore how they're handled by the code in LoopDependenceAnalysis::analysePair, I'm surprised. This loop collects pairs of subscripts from the source and destination refs. * // Collect GEP operand pairs (FIXME: use GetGEPOperands from BasicAA), adding* * // trailing zeroes to the smaller GEP, if needed.* * GEPOpdsTy destOpds, srcOpds;* *

Hi all, After many complaints (including car manufacturers saying the american prompts are unexceptable, EEEK) I started on a quest for real "English" asterisk prompts. The only one I have found is here >> http://www.g7ltt.com/VoIP/vmfiles.html <http://www.g7ltt.com/VoIP/vmfiles.html> And no nothing else on the WIKI looked helpful (e.g. only American voice actors etc)

Hello All, I'm having trouble getting a Zultys ZIP2 to work with Asterisk, along with some other troubles in general. I keep getting a "Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from x.x.x.x). Even when Asterisk reports that the ZIP2 registered correctly, I can't make any calls out from the phone, or calls into the phone. Occaisionally I get a