similar to: Vestec for Asterisk

Does it exist? Sending email to their support address appears to be a black hole. They reference a forum, but Google can't find it. I keep having problems in any grammar than has a an "o" for "zero": it breaks recognition anywhere NEAR it. For example, if I say "two o five", it gets recognized as "one o five".

I'm having a lot of problem with it recognizing "oh" for zero. I've tried both "o" and "oh". In one case, I just tried: $digit = o { out = "0"; } | fundamental {out = "2"; }; So I gave it a choice that was VERY far away from what I said. But when I said "o o o o o", more than 75% of the time, it had a bunch of

Hi listers, I'm a 1.4 holdout who uses mostly Suse platforms. I bit the bullet and installed a Centos box to get Lumenvox up and running, but now see this "new" offering of Vestec that supports OpenSuse and Windows in addition to the Platforms supported by Lumenvox. Anybody out there working with Vestec that can give me a heads up on how it works or doesn't?

Which speed recognition products will also recognize DTMF? In other words, I want to say "Please speak or dial the conference number". Does Vestec allow that? LumenVox? Any other way?

Hi, I would like to see the DNID in my MySQL CDR logs. I have read one big thread in the Asterisk Developer List, but I could not figure out how to implement it ? Is there a simple step-by-step. Thx in advance. Vai

At what stage will there be versions of the G.729 codec, res_cepstal, skypeforasteric, Vestec, etc that'll work with 1.8? It would be good if people using that software could participate in the Beta.

I have a delay of "0" on SpeecBackGround, but when I enter DTMF, there's an almost-exactly five second delay before it returns. Where is this delay controlled? How can I shorten it? Is there a way to set the maximum number of digits to look for?

Hi, does anyone have an info into what could cause [Nov 28 14:24:48] ERROR[11964] utils.c: write() returned error: Broken pipe [Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken pipe [Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken pipe [Nov 28 14:26:23] ERROR[13098] utils.c: write() returned error: Broken pipe Is it a write process or a problem with one

Dear All, I try to be a new user of R. I try to install the R ver 0.65.0 in my UNIX machine. The installation untill one point goes fine but when I run the make comand the program can find the X11 enviroment. Could you help me with this kind of problem? I would appreciate if you have any suggestions about any other process or programms that I need to use. Many thanks for your time.

I have this scenario In the first server 192.168.5.100 I have asterisk installed 1.4.43 and one diguim card with 2 ports: in the first port connection for the provider X : the second port of diguim card the connection of the provider Y In the second server (the same configuration) 192.168.5.200 asterisk installed 1.4.43 and one diguim card with 2 ports : the first port is empty the second

Hi List; In the outbound, I read in the documents the Wildcard match "by using the . (period)", but I did not understand how Wildcard will work (like what)? As I know that Wildcard is a term used with the Diguim TDM card (FXO and FXS), so what is the relation between such cards and the matching in the dial plan? Any help? Regards Bilal

Hi List; If I am configuring Diguim Analoge card, then I need to run #modprobe wctdm, but the question why I need to run #modprobe zaptel also? What #modprobe zaptel does a things that #modprobe wctdm does not do? Any help? Regards Bilal ____________________________________________________________________________________ Looking for a deal? Find great prices on flights and hotels

Hey,?is there any Diguim distributor in Lahore,Pakistan? I need to buy X100P.? Muhammad Faheem -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100326/104853f8/attachment.htm

Ok, if I missed something in the wiki please point me there with the correct search terms. Asterisk 1.0.7 (AAH really) 4 co lines from Bellsouth into a Diguim T400P. Polycom 501 x 4 on the desktops. My problem is on calls to or from the CO I hear a "pinging" (thing sonar ping in a submarine) every 12 seconds. You can set your watch to it. COuld this be a "call recording in

Hello list, 'Just wondering if anyone can point to a very light-weight and easy to incorporate into Asterisk (v. 11.x) to handle a minimal set of responses, like: 0 - 9 yes no (maybe * and # for some people) The idea is that within an IVR menu, the caller could respond by speaking to the typical IVR options, like: For Archie, press or say 1 now For Veronica, press or say

I'm running into some errors I can't figure out. Situation: I have a Sparc IPX (2.6 Solaris), a Mac OS X box and an Ultra Sparc 1 running Solaris 9. I am trying to rsync a web site between the IPX and Ultra 1. All boxes are running 2.5.5 rsync built from source. Problem: Can't get a a sync between the IPX -> Ultra. IPX = kashmir Ultra = vortac Mac = archer Assumptions: a).

Hello to everybody, I have accidentally deleted my cachedmetrics file. I thought that after I run wine and the file is not present, wine would generate it - but it didn't :( Does anybody know how can I generate that file? Thanx in advance Stelios -- http://www.freemail.gr - ?????? ???????? ???????????? ????????????.

Can anyone recommend CRM software with a link into Asterisk? I would like a pop up on caller ID if possible. I've played with the FOP and SugarCRM but can't get them working together. Regards Lee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/e71c34a3/attachment.htm

Is it true that asterisk 1.2.6 does not use mpg123? I just installed asterisk 1.2.6 and while I do have music on hold (through format_mp3?) I do not have an mpg123 process running. I seem to be having serious audio issues when going through one of my providers (and just through that provider) when using mp3 for hold music, however when using wav files it is fine. The processor is only at about

Hi, I' ve just connected a carte X100M to my asterisk server running zaptel-1.2.5, libpri-1.2.2 and asterisk-1.2.6 on SUSE 10.0. When I make modprobe wcfxo and modprobe zaptel I haven't any error, I have also chan_zap.so module existing in /usr/lib/asterisk/modules. But, when i run ztcfg, it shows me this: Zaptel Configuration ====================== Channel map: 0 channels configured.