similar to: How to use * to simply skim off callerid (UK)?

Is there any way to take an incoming callerid string and remove the given "name" part of it and replace it w/ something arbitrary, or add to a blank name string (possibly by looking up the number in a database)? Thanks, John Lawler

Please can someone put me out of my misery... I am missing the point here: I have a table of albums called: dealers I have a table of images: popp_images each dealer has 6 popp_images linked by foriegn key: albumid Each dealer can login using user_engine engine I restrict viewing for each dealer to their own images by assigning each dealer their own albumid. I list their images with: in my

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Greetings, I need if possibile an explanation on how to manage the incoming callerid for an incoming call. Let me explain the situation: We have two different companies in this office that shares the same PBX (* box). Each company have its own number for the incoming calls. What i'd like to implement is something that, depending on the incoming line that is involved in the call, plays a

Dear list subscribers, this is posted in appreciation of the level of skill that subscribers to this mailing list enjoy and in no way to abuse it. We are looking to hire a Research Developer with extensive R and/or python development skills. If interested (below), please contact me directly on osklyar at ahl.com: Man Group / AHL Research Developer [3040] Man is a world-leading alternative

Hi guys, I'm running Asterisk-0.5.0 and accidentally stumbled on this problem while in the VoicemailMain2 application: If you login to it, or even if you call it w/ 's<extension>' to skip the login and press an '8' near the beginning (and possibly at any point, I'm not sure), the channel seems to lockup, even if the handset is hungup, the channel remains frozen

> > > >| -----Original Message----- > >| From: asterisk-users-bounces@lists.digium.com > >| [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > >| Michiel van Baak > >| Sent: Martes, 24 de Mayo de 2005 04:45 p.m. > >| To: asterisk-users@lists.digium.com > >| Subject: Re: [Asterisk-Users] CallerID > >| > >| On 23:44, Tue 24

Dear Colleague, I am writing to ask your help for a book I am writing about Life Practices. I am looking for examples of those simple things that we all do that are essential to expressing and maintaining our values as we face life's ups and downs - our Life Practices. What I am seeking is a bit like folklore. Something with a story and an easy to remember slogan or catchphrase. I am asking

Hi guys, I posted a somewhat similar question about a month ago and got a thoughtful resonse from Steven Critchfield, but I've got a quick follow up question to it. I'm looking to setup a 16 extension / 10-14 phone line Asterisk install for a customer who would like to have DID numbers for the extensions, since they're currently on Centrex and already have the 1-to-1

Trying to revive this old thread as problems continue. I have noticed from the gluster logs the following on m y volume called export: [2017-08-16 20:08:47.663908] I [MSGID: 109066] [dht-rename.c:1608:dht_rename] 0-export-dht: renaming /projects/ACTIVE/Automotive/JEEP/Brand Image Program June 2016/04_Western Region/Huntington Beach CDJR - Huntington Beach, CA/04 REVIT AND CAD/2017-08-16 CAD dwgs

We are using Voicemail 2 and use the e-mail with WAV file attachment option. When we record a message from an internal extension, the WAV file sounds fine. However, if a caller from outside leaves a message, the volume on the WAV file is so low it is almost useless. When we access the voicemail through an extension, however, the volume sounds fine. Has anyone had this problem or any idea of a

On Tue, January 27, 2015 1:58 pm, Peter Lawler wrote: > On 28/01/15 04:47, Always Learning wrote: >> >> Saw this on the Exim List:- >> > <SNIP> >> >> I use Exim on C5 and C6 - should I be worried about Exim on C6 ? >> > > upstream references: > https://rhn.redhat.com/errata/RHSA-2015-0092.html When I read this I read that it is fixed in

I'm trying to get musiconhold to work w/ my Asterisk system, and I'm not having much success yet. First, I noticed that nothing happened even after I had enabled all of the options in zapata.conf & setup a sample extension in extensions.conf. Then I read something about how Asterisk uses mpg123 to play the files. I discovered that this had not been installed on my system, so I

Hi guys, Thanks for your answers on my two questions yesterday. That's exactly what I was looking for, sorry for not noticing it myself, but I'm still getting acclimated to Asterisk and even Linux--from what I see so far, I love it. I've got another one now. Since my Asterisk install and configuration is fairly stable at this point, I'm interested it ensuring that during

On Tue, Jul 13, 2021 at 8:20 AM Jeffrey Osier-Mixon <jefro at redhat.com> wrote: > > Hi all - I'm able to log in, but not yet able to edit the SIG proposal. Would it be possible for me to have access to this doc? > > https://wiki.centos.org/SpecialInterestGroup/Automotive > > thanks > > Jeffrey "Jefro" Osier-Mixon | jefro at redhat.com > Red Hat

Hi all - I'm able to log in, but not yet able to edit the SIG proposal. Would it be possible for me to have access to this doc? https://wiki.centos.org/SpecialInterestGroup/Automotive thanks Jeffrey "Jefro" Osier-Mixon | jefro at redhat.com Red Hat Office of the CTO | Sr. Principal Community Architect, Automotive On Fri, Jul 9, 2021 at 10:08 AM Jeffrey Osier-Mixon <jefro

Hello, llvmdev. I'm a student from the Moscow State University and I started to work with the LLVM project. First, I have to compare its profiling ways and I got a little problem. It hangs when I try to run optimal edge profiling. Nothing happens right after the command "opt a.bc -profile-loader -o b.bc" which tries to load optimal edge profile from the llvmprof.out file. I mean it

Thanks everyone for your feedback, especially Mark. I now have the ALL the files I need. My order still stands for the $8.00 product from CISCO but the CP7960 dealer sent me all the files. Now I will move on to completeing the setup of the TFTP server. Thanks again -----Original Message----- From: Michael Loftis [mailto:mloftis@wgops.com] Sent: Friday, February 18, 2005 7:51 PM To:

G'Day All; So I purchased a Cisco 7960 and am now trying to get it configured for *. No can do without the variuos files/images through a FTPF server. I configured the TFTP server on my RHES 3 box, now to get the required CISCO files. So I contacted CISCO to purchase the required maintenance contract so as to gain access to the download area for the files/images. -WHAT A FRUSTRATION!!-

Hi guys, I just moved from Asterisk release 0.5.0 to CVS 2003-12-22, and after overcoming a few changes in my configuration, I encountered one problem that I couldn't shake that was working fine in 0.5.0. It's the fax detection. I just have a simple extension setup like this: exten => fax,1,Dial(Zap/4,30,tr) exten => fax,2,Hangup in my main incoming context. This used to