similar to: ISDN PBX + IVR + Voicemail Configuration - S anity Check ...

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

Hi All, First of all, please no flames about the footer of this message. It's being inserted into all outgoing email by the server here without asking me personally first once I send out email. Since I've been very interested in setting up an Asterisk PBX + IVR + Voicemail system to support a small business and a foundation for quite some time now, in order to replace their current ISDN

Kapejod is working on a fix for the CDR problem in bristuff. See below ----- Forwarded message from kpj@junghanns.net ----- Resent-From: tzafrir.cohen@xorcom.com Resent-Date: Fri, 10 Feb 2006 13:01:25 +0200 Resent-Message-ID: <20060210110125.GU16880@xorcom.com> Resent-To: tzafrir@cohens.org.il Envelope-to: tzafrir.cohen@xorcom.com Delivery-date: Fri, 10 Feb 2006 05:19:50 -0500 Date: Fri,

All, I would like to play an announcement to the user on what external line a call came in, right before this call get bridged to this user. How would I go about implementing this in * ? Regards, Hans -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may

right, so far, here is what I've done: I succeed in take in a new variable the UUS1 field sent with the connection request for incoming calls. It was quite simple afterall... (I just had to find where the data CMSG->Useruserdata is coming in chan_capi.c) Now I would like to know where this field is instanciated for outgoing calls in order to control this step? I am looking for that but I

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Hello everyone, ff123 compiled Monty's branch of the RC1 encoder, see his post on r3mix.net forum: http://66.96.216.160/cgi-bin/YaBB.pl?board=c&action=display&num=994299736&start=30 Anyway it only supports ~128kbps mode, so I did a quick listening test with some files that bugged vorbis beta4. grace.wav - the right channel is still a bit watery, and I think this can be seen

Hi, I receive the following warning message when I run a poisson GLM in R: "glm.fit: fitted rates numerically 0 occurred" The model summary is shown below. The variable 'Species' consists of counts of different species ranging from 0 to 4. I suspect this may have something to do with the warning message but I'm not sure. Can anybody help? Thank you! Anna Call:

Guys, I am planning on building a small SIP PBX with a single ISDN line. Currently I am looking into the specs of a very tiny barebone system that has an option Colognechip base ISDN controller. The only thing is that the ISDN module that comes with this barebone hooks up to the motherboard using USB. My intention is to allow incoming and outgoing calls from SIP to ISDN. Is this setup in any way

These allow and disallow work with NuFone for me disallow=all allow=ulaw allow=alaw allow=gsm Jeff Message: 11 Date: Fri, 11 Mar 2005 11:15:51 +0100 From: "Edward Banfa" <edward@radform.com> Subject: [Asterisk-Users] NuFone Configuration [problem] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com>

>>Those things generally happen in Amsterdam. And in Kristiania in >>Copenhagen. The usual problem: Smoking too much pot I have to object to that, as a rule of thumb the Dutch only rob tourists who are dressed like tourists and act like tourists, that's what we all agreed to here and live by -- please just dress local and act local, so we can finally stop smoking pot just to keep

>>>Those things generally happen in Amsterdam. And in Kristiania in >>>Copenhagen. The usual problem: Smoking too much pot > I have to object to that, as a rule of thumb the Dutch only rob tourists who > are dressed like tourists and act like tourists, that's what we all agreed > to here and live by -- please just dress local and act local, so we can > finally

Hi all, i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco ATA-186 3.1.1 atamgcp We are used to make an special ;) blind transfer like (Flash)Number(Hangup before anyone answers or ring). Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp If one waits until the last one rings, then hangup, everything is fine. If one waits until the last one

Hi, BRIstuff contains two bugs in its implementation of the Redirect manager action: 1. If the property ExtraUnqiueId is used, the Priority property is used to redirect the extra channel (instead of ExtraPriority) 2. If the property ExtraChannel is used, 0 is used to redirect the extra channel regardless of the Priority and ExtraPriority properties. A patch for manager.c is available at

Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on

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I have tried many permutations of search terms in google and I haven't been able to find any confirmation that the megaraid* modules in centos4 supports port multipliers/multiplexers. Here is a link to the controller I have in mind at newegg: http://www.newegg.com/product/product.asp?item=N82E16816118032 In case anyone is interested, here is what my research has turned up when looking for

After apt-get update && apt-get upgrade -y wget http://www.junghanns.net/asterisk/downloads/chan_capi.0.2.5c.tar.gz tar xfvz chan_capi.02.5c.tar.gz cd chan_capi-0.2.5c make && make install shutdown -r now asterisk seg faults upon calling in via ISDN. Any ideas are greatly appreciated. rgds pos

i want to tag categories to its menuname. i have a csv containing menu item name and in other csv i have a column containing some strings, i want to pick that strings from categories and look into menu items if any menu item containing that string i want to create a new column next to menu item name flagged as 1 otherwise 0 and the only condition is once a menu item flagged as 1 i don't need