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Hi all, we are considering Asterisk as a possible solution for providing VoIP services. The list of requirements would be: 1) Load balancing and clustering (horizontal scalability) 2) User portal for self provisioning Googling around doesn't yield any "standard" state-of-art solutions for points 1 and 2. Is anyone using asterisk for this scenario? Which solution are you using?

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

Hi, I have problems with dovecot + ldap + nfs, we could't have more than 20 concurrent sessions of pop3, when user 21 comes, the auth process takes more than 1 sec and is incresing (while most users arrives, more delay in autenticacion exists) and my customers complaint about the service. We need support about 200 concurrents pop3 users, here is my details Front Internet 1 Load Balancer

Hello all, I haven't used cisco phones as yet - do they work with asterisk, are they good which models are the best? I am after a starting point! Thanks Nick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040422/f5246238/attachment.htm

It's been down the last 5 hours at least. Anyone know what the problem is, or when it will be back up?

Is there anyone with any experience of using the AMD app and the settings that worked for them in the UK ? Any help would be appreciated. Julian

For all who are interested: A quick review of CeBIT 2005. :-) CeBIT was a very successfull event. Most of the time, the asterisk-booth was crowded with more people than we could talk to. We had with us a demo-installation including different IP-phones, digital and analog phones as well as a Siemens HiPATH PBX to which our Asterisk-server served as a VoIP-gateway, and many people were impressed

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

Hello list! I?m working on a fresh Asterisk install over CentOS7 base. I?m using ?Asterisk. The Definite guide? book as a reference. I connect and work using SSH Problem I have - I can?t connect to asterisk from remote. Getting error: $ sudo asterisk -rvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Yes, it exist, and service runs: [asteriskpbx at

It would be nice if Digium updated the hardware compatibility list on asteriskpbx.org to indicate that the LineJACK can't be used for dialing out. I've seen several people on IRC be burned by not knowing this. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Dear, I've installed the following R version on HP compaq 6710 b under Fedora 8: > version _ platform i386-redhat-linux-gnu arch i386 os linux-gnu system i386, linux-gnu status major 2 minor 6.2 year 2008 month 02 day 08 svn rev 44383 language R version.string R version 2.6.2 (2008-02-08) When I try to install some R packages I receive the following message: [root at localhost

Hello folks. I've got a strange issue. When I modprobe TE220 I do not see mesages like Launching card: 0 <..> Setting up global serial parameters. You can see how I loaded and unloaded the card for several times - http://asteriskpbx.ru/pastebin/11 lspci can detect the card: 03:08.0 Communication controller: Digium, Inc. Device 0220 (rev 02) dahdi_hardware also: astpbx ~ # dahdi_hardware

Hi list, To make outgoing calls by skype i would like to have our crm app create callfiles like we do for normal calls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[<originator>@]<destination>) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228

Hi Folks, Does anybody have experiences with plugging 3 TDM400P cards in one PC?? I think about a Asterisk box handling 8 incoming analogue lines and providing 4 lines to an old analogue PBX. I read a lot about trouble with the TDM400P cards so this idea seams to be not really god, or? Ciao Joerg -- _____________________ Don't PANIC

Buenos días, estoy haciendo unos mapas con ggplot, con 29 categorías, por lo que tengo que utilizar library(RColorBrewer) para disponer de suficientes colores. El problema es que al hacerlo acorde a dos variables distintas (color=var1 y color=var2), cuyas 29 categorías son obviamente las mismas, les da distintos colores y no puedo comparar los mapas. Muchas gracias, Manuel

I am able to get this script to dial, but I am unable to talk or hear anything. The script asks for the number to call and the the caller id to display (if user is not at their normal extension). Once submitted, the external extension receives a call, once answered the call is then placed to the dentition number. The script works as the call is place, but I cannot hear or say anything. Any one

I have a gateway, which I call from my mobile phone (free of charge, since it is the same phone company). This gateway gives me a dial tone. I can than dial to any extension number or even other gateways, .... It is getting more a trouble to remember all the numbers, or to key in all the long phone numbers when you got the dialtone. I was thinking of using for this Sphinx2. How can I

Hi I was doing some bandwidth testing, and my incomming usage is 36% more than my outgoing bandwidth. The setup is IAX2 trunking using GSM codec. Is there any obvious reason I am overlooking to figure out why there is such a big difference between the two.? I am using CVS-head September 3rd, maybe there is a version skew? Any suggestions will be appreciated. Thanks Clive

This may sound kind of crazy and I maybe missing something. But are you placing the call on hold so you can hear the hold music. This may not be the case but you may have to place the call on hold to here the music. Also you mentioned sound, you do not need a sound card in the asterisk box to use this hold music feature. Hope this helps. -----Original Message----- From: