similar to: Asterisk & iSeries AS/400

I just got cepstal working fine in the dial plan using code like: exten => 511,5,AGI(cepstral.pl|Welcome to my house finder. At the beep enter your zip code.) The php script it calls is based on the nerdvittles weather one so it calls a webpage which prints to the screen, the nerdvittles code uses system to generate the .wav file then has the dial plan call it via: //php script $retcode2 =

are there any work in progress for this platform (iSeries) ?? thanks Hugo -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20060510/a968bec2/attachment-0002.html>

Hi, Has onyone heard of successful deployment of 3Com ToIP over IBM iSeries system (formely AS/400) ? A prospective customer seems to looking for this but, in my whole life, I've never of a such setup. Does it work ? regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm using Cepstral as a TTS Engine for Asterisk with Swift application. It works fine when I have just 1 voice installed. Now I have 2 voices in the same language installed but I can't seem to find the way to select which voice to use in Swift's application in Asterisk. Does anyone know?? Thank you, Pim

Hi all, I hope that I'm not breaking protocol too much by posting a message in this group about a problem that I'm having with an Asterisk Business Edition installation, but the reason that I'm posting here is because the problem that I'm having isn't really with the Business Edition, it is with the Cepstral text to speech product that I'm using with it, and also because

I'm using trixbox and the asterisk agi. I downloaded a cepstral voice and worked with it until I got the code to do what I wanted. I then registered the voice today to get rid of the 'this voice is not yet registered", stuff yet it still does that. Any ideas on how to fix this? It told me my info was valid. Thanks! -------------- next part -------------- An HTML attachment was

Hey everyone, I was frustrated with the existing app_cepstral/app_swift TTS modules I've found on the net, so I hacked up my own. It's been working really well for me so I thought I'd share. In developing this, I wanted to avoid: * the startup delay incurred writing TTS output to a temp file then playing it back * gluing cepstral/swift to the asterisk module in a way that fubard

Can you say a Virtualized Asterisk with a PRI card! http://www.phoronix.com/scan.php?page=news_item&px=OTY0OQ Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

>>Your membership in the mailing list asterisk-users has been disabled >>due to excessive bounces The last bounce received from you was dated Anybody else seeing this? My mail server logs don't show any issues. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

Last month I moved a 1.4.x Asterisk install to Asterisk 11.5.1. Everything is working well, until I noticed that Caller ID between facilities are showing properly, on the phone display, until the handset is picked up, then it's displaying NAME NOT FOUND. I do database lookup against extension number (Of remote PBX) and use that as the MySQL key to pull name. Snippet below: -- Executing

I am regularly running into a problem where I can't seem to figure out how maintain correct data order when selecting data out of a dataframe. The below code shows an example of trying to pull data from a dataframe using ordered zip codes. My problem is returning the pulled data in the correct order. This is a very simple example, but it illustrates a regular problem that I am running into.

We currently have an Asterisk system that is hooked up to our old paging speakers via sound card, plugged into two amps. Each amp drives up to 8 analog speakers in each warehouse (we have 2). Both warehouses are around 30k square feet. Both have a large number of printing presses. The computer system is that is running Asterisk is around 10 years old and starting to fail. I'm looking to

please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.......................... On 4/10/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing

I know this is probably really easy thing but... I've loaded data into R from the net. I've got 97 subjects (in a database). I'm supposed to remove subjects "lower than 140 cm".. Of course it's easy to find the subject (i.e. number 83 and 84) - but how can I delete these subjects from the database in R? (delete the entire row..)? I've spent 2 hours searching the net

Hello all, I am trying to get cepstral to work with Asterisk for Text to Speech purposes. I got Asterisk to recompile with cepstral and after doing *CLI> show cepstral female voices Available Cepstral voices: Name Gender Age Language Sample Rate License ========== ========= ===== ==================== ============= ============ Amy female 30

Hey everybody, I've, within the last 3 weeks, moved over to a PRI from SBC/AT&T. I've received several complaints about dropped calls. Reviewing the archives on PRI and dropped calls shows that I should set the resetinterval=never in the zapata.conf and restart. This hasn't helped. The dropped calls have to date only been on outbound calls. Usually within 2 to 3 minutes

Quick question. Say I have a geographical database with counties and zip codes where counties have and belong to many zip codes. zip_codes (id, zip_code) counties (id, name) When I create the association table, the Rails way says to do the following: counties_zip_codes (county_id, zip_code_id). However, given that zip_codes.zip_code is itself a candidate key, I would much prefer to do the

Some of you may know me as ManxPower from #Asterisk at irc.freenode.,net I've posted my demp weather report Asterisk AGI script at http://www.fnords.org/~eric/asterisk/downloads/ I have no affiliation with Cepstral. Below is the README: Contact: Eric Wieling <eric@fnords.org> If you want a demo of this AGI script you may call via IAXtel 1-700-923-3645 extension 2101. Option 23 is

2013/2/12 Doug Lytle <support at drdos.info> > >> non-standard locations such as /usr/local/sbin > > If compiling from source, it'd normally be specified by the --prefix > option: > > ./configure --prefix=/usr/local > > Doug > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary >

I don't suppose anybody has an updated CalledID patch for Asterisk 1.4.26: https://issues.asterisk.org/view.php?id=8824 I've been running 1.4.21.1 for a while and have the need to update one of our Asterisk Conference/Faxserver/switches and wanted to see if the patch available would apply. It failed in several places. This keeps me from updating my other installs. Doug -- Ben