similar to: PhoneCALL v2.6.1 - 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

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

Greetings Everyone! The version 1.0 of the PhoneCALL Administrative Manual has been released. It is more of an outline of the features and interface, and we'll be adding lots of more detailed information in the manual over the next few days/weeks. Of course, we'd love to get your input on the manual and areas we need to clarify or even some new sections in the manual that would help

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

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

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

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 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

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 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!

Hallo, we have started playing with asterisk about one month ago, and we do like very much what we are experiencing. Now we would like to take some step further towards "standardizing" installed modules, functionalities, tools etc. The "wall" we are facing now is: choosing the right tool for * management. We tried AMP, very powerful but incomplete (CAPI is very important to

Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager image? I want to start playing around with the chan_skinny addition, but it seems the .exe's from cisco want to open a connection to a SQL server or CallManager (which I don't have).

Is there a way to get Music-on-Hold to use the Line input of the soundcard instead of playing MP3s? Some potential clients like the music-on-hold to be the local radio station, or from a satellite broadcast. Thanks!

This is Off-Topic for Asterisk, but I wanted to get some feedback on headsets for Cisco 7940/7960 phones. We have about 10-20 people who wants/needs a headset for their phone & was hoping to collect some real-world input. Thanks!!

I would like to add the ability for our users to be able to press "0" whenever reaching someone's voicemail box to re-reroute them to the auto-attendant. Here's a sample extensions.conf: [incoming] include => ciscophones exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten => s,5,BackGround(auto-greeting)

Hello~ When I use R2.6.1, the following work correctly. par(mar=3Dc(4.2,3.1,0.5,0.5), oma=3Dc(2,2,2,2)) boxplot(count ~ spray, col =3D "lightgray") box("inner", lty=3D"dotted", col=3D"red") box("outer", lty=3D"solid", col=3D"green") =20 But when I use R2.7.1, the outer region seems too large. I don=A1=A6t know what

I would like to write AGI script in Ruby Would anybody please show me right direction.. Thanks

I have a client with 20 GXP-2000s. Everything seems to be working fine. However, after a couple of weeks of use, the client is having a hard time adjusting to the new IP based phone systems and only misses one feature from their old Lucent system. That is, they had 8 analog lines before and all their old Lucent phones showed a button for each line. So, it was easy for anyone to say,

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

A customer of mine wants an IVR where the first 3 choices are 1 English 2 Spanish 3 French I can build the IVR but how do I get the system prompts to then speak the selected langauge. For example, a caller has selected Spanish and so is routed to the Spanish part of the IVR. At some point he breaks out of the IVR to leave a VM. How does the system know to continue offering him Spanish?