similar to: PhoneCALL version 1.0 Administrative Manual - Released

Hello all! It's come time where I need to add another programmer to our team. You should have at least 3 years of "work" experience with PHP/MySQL. Please send me your resume and a few code samples if you can. If you can only work part-time or full-time, please include that in your response. Along with your salary requirements. You'll be working with PhoneCALL, so be sure to

Hello All! Just a notice that our PHP/Smarty-based GPL version of PhoneCALL version 2.6.1 has been released, and is the current stable release. http://www.vecsector.com/phonecall We're always looking for feedback/testers to help us enhance it and make it even easier for everyone to use. The current version is designed around the advanced Asterisk user, and we are working on a more

Has anyone every heard of Phonecall? : www.vecsector.com/phonecall/ Feedback? Is there something as good as it or better ? Recommendations? -- ================================= Joshua Abbott, Support Technician http://www.successfulhosting.com/ Direct Line: PENDING Phone: (866) 494-5096 x1207 E-Fax: (419) 858-3241 Alt E-Fax: (801) 217-1123 jabbott@SuccessfulHosting.com

Hi all, I need to use asterisk to record all phonecall I have test using mixmonitor to record a call. Now I need to set the configure file to let asterisk auto record all calls. I have searched many document but still can not succeed. My version is 1.8beta and I prefer using mixmonitor. Regards!

Hello - I am working on creating an Asterisk call center queueing application which will be an addition to our hosted services product. I currently have the functionality I need developed within Asterisk, but am falling short on finding a solution to provide the customer with a user interface into their queue statistics. I have evaluated queue-metrics and Asterisk-Guru's queue statistics

What third party programs are available for parsing the queue_log file and CDR file? I know about XC-AST, but management would prefer a php based solution. What have other admins done to retrieve detailed call information about the queue system? Anyone develop their own that they don't mind sharing? --johann

I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks

I got it figured out now. In case anyone else comes across it - this is what I had to do: Sethdlc hdlc0 mode ansi Ifconfig hdlc0 up ***NO IP INFO SET*** Sethdlc hdlc0 create <dlci number> Ifconfig pvc0 <local ip> pointopoint <remote ip> up Then set your default gateway to the <remote ip> listed in pvc0. That's about it!! Hope it helps! -----Original

I wasn't thinking of using the conference system as the basis. I was thinking more along the lines of: 1) Setup a second extension on the Cisco phone named "INTERCOM" enabled for auto-answer 2) Create a call group on asterisk to dial that "INTERCOM" extension on every phone that will participate 3) Add a feature code that would dial the intercom extension and connect

Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Does Asterisk BE come with a GUI management console for managing phones, queues, VM and the like? -Matt -- Matthew S. Crocker Vice President Crocker

Hi. I've started working on a PHP-project that generates the configuration files i need based on what's in my MYSQL database. I can add, delete and edit users from the web. I can set up exactly the dialplan i need by arranging the users in a firms and groups if needed. I've also set up a java servlet so that i can get asterisk to reload by pushing a button from the web-interface. The

Hi there. For some reason I am getting permission denied errors on a new samba server. I can add machines to the domain, I can login with any user id's and so on as well. But when I try connecting to a public share, or the users home directory, I get a user denied error. "\\Fileserver\user is not accessible. You might not have permission to use this network resource. Contact the

FYI -------- Original Message -------- Subject: RE: AOL Mail Date: Wed, 25 Jun 2008 13:47:06 -0500 From: Dustin Davis <dustin@txls.com> To: "'Dale Schroeder'" <dale@BriannasSaladDressing.com> References: <419cf800ca3991a24f4b9829f3806aba@briannassaladdressing.com> <145401c8d604$ccfb1780$66f14680$@com>

I have some problems with busy detection and SIP. When I'm making a phonecall (out or internal) and someone else is calling me, the phone (Snom200) is ringing and leaving the first caller (no difference if I call someone or if someone calling me) in the background waiting. It doesn?t hang up the first call but the second one is overriding the first. Is there anyone that has experienced the

Kathy, Thank you very much! Peter Olson Alcatel-Lucent Member of Technical Staff Services Technologies 1960 Lucent Lane Rm 7G334 Naperville, IL 60566 peterolson@alcatel-lucent.com Phone: 630 979 0573 Mobile: 630 430 6926 -----Original Message----- From: xen-users-bounces@lists.xensource.com [mailto:xen-users-bounces@lists.xensource.com] On Behalf Of xen-users-request@lists.xensource.com Sent:

Hi I was given the task to try and build a VOIP solution to an office building with multiple tenants in all sizes and shapes. Some of them will require auto attendants and some will simply want direct lines to their phones. The question I have is: Can asterisk be configured to handle multiple auto attendants? TIA Paolo

Hello Dustin, Always inline is the closest to a preprocessor macro you can get in LLVM Assembly since it doesn't have a preprocessor at all. LLVM does aggressive inlining for functions used only once so those instances don't require specification as alwaysinline. --Sam ----- Original Message ---- > From: Dustin Laurence <dllaurence at dslextreme.com> > Cc: llvmdev at

> -----Original Message----- > From: Uwe Klein [mailto:uwe-klein@foni.net] > Sent: Monday, January 19, 2004 9:14 AM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Compiling problems with SuSE > > > > From: Dustin Knuttgen on Sunday, January 18, 2004 11:47 PM > > > > > We tried to use SuSE initially and had no luck compiling zaptel

I am having some problems with uploading printer drivers. I follow the procedures as outlined in both TOSHARG and Samba-3 by Example, and I use the Add Printer Drivers Wizard to upload the drivers. It looks OK, and the drivers make into the \\server\print$\W32X86\3 directory. But when I click okay, I get "Printer settings could not be saved. Access is denied." This happens when

Dustin: I am adding this response to the libvirt list as well so that others may see it. The authorization is done via a callback. The ConnectAuthDefault() that you are using is one such implementation. You could create your own subclass of ConnectAuth, and override the method: public abstract int callback(Credential[] cred); And then provide the username/password in that fashion. Look at