similar to: Professional Recordings

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

Hello, New subscriber here. I noticed that the FTS index is not used in compound searches. Is this expected? Tested in 2.0.0 and 2.0.8: . search BODY "waldo" * SEARCH . OK Search completed (0.000 secs). . SEARCH CHARSET UTF-8 OR SUBJECT "waldo" FROM "waldo" * SEARCH . OK Search completed (1.768 secs). . SEARCH CHARSET UTF-8 OR SUBJECT "waldo" BODY

Hello, I have an asterisk server running with 23 g.729 licenses. I have also purchased a sound file from thevoice.digium.com. I need to covert this file (uLaw, PCM I think) to g.711, g.729 & g.723 for use with an IVR system. Is there a way I can convert the files using the g.729 digium codec? sox? Thanks -Matt -- Matthew S. Crocker Vice President Crocker Communications,

Any chance of say digits being extended to recognise "*" & "# " ?? Heck these are digits on a normal keypad :-) Gary .

I'm looking for an inexpensive way to connect 20 analog phones to asterisk. I could get a bunch of Linksys or Sipura boxes but was wondering if there is a more cost effective way? I came across the Mediatrix 1104 and even the Mediatrix 1124 but that comes out to be almost $100/port. I might as well buy inexpensive IP phone. Does anyone have any suggestions? Thanks, Waldo

Hello list, We have some kind of a problem with our Asterisk installation. We want to be able to publish different caller id when placing outbound calls through the PSTN. We have Asterisk with TE410P and T1 from FDN Communications. The problem is that all our outbound calls show our main number, regardless of what we set with SetCallerID, even using CallingPres with all possible

I'm planning on setting up some remote agents and before doing so, I did some simple PING tests to measure latency. The average latency I got was 250ms. Does anyone have experience in terms of quality of calls when there is such high latency? Can anyone comment? Thanks, Waldo

I'm wondering if anyone can recommend one over the other. I'm mostly interested in running open source solutions, so I would prefer if your recommendations are within the open source arena. Basically, I contemplated the idea of using SER as a NAT Helper and possibly as a SIP server for a portion of our user base. We prefer to have Asterisk in the mix because of the additional

I am the tech coordinator in a small public school in rural North Dakota and all of last year I ran Samba on redhat 8.0. I had XP machines that used Samba as the PDC. Worked fine. But now I am getting some XP home machines that cannot log on to a domain. That's okay, because I am determined this year to use solely redhat. In fact, I put the partitions down to 5 gig for Windows and

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 all, Assuming 1.0.7 is the latest stable version, how/where can I find out the different CVS revisions available and a description of what has been patched/updated in each CVS revision so I can decide whether to leave my 1.0.7 installation as is, or if I need (or think I need) to patch it with a CVS version? Thanks, Waldo

I was browsing the applications developed in zaptel and came across zttest. After I run it, I get the following: Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.000000% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%

I was playing today with the different queueing strategies in queues.conf when I noticed the following behavior. I have 4 agents defined in a queue in queues.conf. These agents login using AgentCallbackLogin. The strategy in the queue is set to leastrecent. I place four calls into the queue and * sends only one call to the least recently used agent. If that agent does not pick up, the

Hi list (again), I have another question which I have not been able to resolve from neither the wiki nor Google. I've been able to configure a multi-tenant setup of asterisk for 2 small call centers with no problem, by simply playing with contexts (which I guess is how everyone else is doing it). The problem I have is that I've only been able to configure one global agents.conf

I have two asterisk machines. One of them has a Digium board (server A) and the other is simply using ztdummy (server B). Server A is running on Debian and Server B is running Gentoo. Server A is running Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 and Server B is running Asterisk 1.0.7. The problem I have is that when I try to transfer a call into a meetme room in server B, it simply hangs

Hi We are trying to get better sound quality out of the prompts on our Asterisk system. We had some new ones made by thevoice.digium.com and they are in WAV format instead of the default GSM format on the Asterisk server. The problem is, when you dial in to the server using the FXO card (X100P) you really can't tell the difference between the WAV prompt and the GSM prompt, however, if you

I've been looking at the weather vocabulary in asterisk-sounds in CVS. I've run into a few hitches with words I can't seem to find. So far, I'm looking for 'point' (for constructing floating point numbers) and 'around' as in "high around 70" (don't I wish). Any chance of getting these? While I'm on the subject, I'd be very interested in a

Hello everybody I am looking to push out httpd24-httpd-2.4.25-9.el7 to my organization, but I do not see it as being available on the mirror.centos.org site. I see a git commit for this package in April and was wondering how long it takes an rpm to become available once the commit has been completed. Also, I don't see the following CVEs addressed in any httpd24 changelogs and wanted to know

Hi, I have a lot of GXP-2000 phones not registering with Asterisk server. After two days of attempts, it seems that problem is due to the fact that phones and server are not on the sme network. Do you know if this is known issue? -- Domenico Viggiani

before I accuse somebody to "overbill" I would like you to calculate with me: Rate: 0.0189 for calling Taiwan via NuFone Duration: 930 seconds Lets vote for the answers: 0.7269 or 0.2929 ??? bye Ronald Wiplinger