search for: phonecalls

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

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

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

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!

Hi, I've got a (maybe stupid) question: I'd like to install Asterisk on my notebook (just to play around with it). Is my internal phone modem equivalent to an FXO port? Hence, could I make phonecalls to my pots-line with it? thanks philipp

Hi, I've got a (maybe stupid) question: I'd like to install Asterisk on my notebook (just to play around with it). Is my internal phone modem equivalent to an FXO port? Hence, could I make phonecalls to my pots-line with it? thanks philipp

Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [root at asterisk asterisk]# cat

hi everybody, I'm currently planning a small-sized web-applicaiton allowing users to call-in via phone. the phonecalls should be recorded and processed further by some custom scripts - sounds like asterisk is a perfect match for this app. however, during prototyping I have no ISDN-connection whatsoever available, so I was asking myself if it's possible to connect a cellphone via data-cable (or bluetooth?) and...

Hi, I created a modification (more like a hack) to shorewall that backups a configuration after succesfully (re)starting it. In case a future (re)start fails it will use this backup configuration. (instead of stopping the firewall and generating a massive ammount phonecalls ;) I didn''t pay too much attention to the ramifications of this patch, so let me know if i screwed anything up. It seems to work well with our Webmin administrator interface. Edwin -- //||\\ Edwin Eefting || || || DatuX - Linux solutions and innovations \\||// http://www.datux....

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

hi, im complete new with asterisk, so.. i want to be able using dtmf during a call, for execute a application. Now i'm still making phonecalls trough a sip-adapter (Linksys pap2) with my soundcart from server, and i receive in the asterisk-console putting some digits from a analog-telephon:" << Console receive digit 1 >> " e.g. Now how can i execute some application with this digit?. I have tried, to execute with a...

Hello, Just confirmed this on my end, because of the massive changes that have been made to callerID handling in asterisk 1.0.5 many of the features of the astGUIclient suite will not work on this new version. The latest stable version recommended is Asterisk 1.0.3. We will work on trying to find ways around the new callerID rules that the asterisk developers have put in place and hope to have

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

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 all, I am struggling with the following and I can't get it work: In the Netherlands where I live it is possible to use special codes (aka vertical service codes) to set special 'behaviour' of phonecalls. So e.g. when I want to dial out with a normal phone and I dial *31*<phonenumber to dial> then it will turn off my numberindication (CID) at the called party. They seem to call this the 'keypad protocol' but I cannot find this term when searching through asterisk documents. My aste...

I have been doing some reading about gtalk and asterisk. Most of it is pointed to enable using gtalk for making phonecalls. Would it be possible to use gtalk instant messaging input (just some text send to the gtalk account configured on an asterisk box) into the dialplan. This way you could use gtalk im to trigger all kind of events like sending an sms, adding sip entries to the system, start conferences etc...

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

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