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Hi, I have these extensions: exten => 101,1,Dial(SIP/101,15) exten => 102,1,Dial(SIP/102,15) exten => 0,1,Dial(SIP/101&SIP/102,15,r) They work fine and I get the ringing sound if I dial them directly. However, I also have this extension: exten => s,1,Answer() exten => s,2,Background(viagenie) exten => s,3,WaitExten() The ringing sound doesn't work for any extension

Dear people, is it possible to have presence using Asterisk 1.2.13 / SIP with Linux/Debian Etch??? I'd like to see if my intranet contacts are available, busy, disconnected.... Thanks a lot Alejandro

hi, can you recommend "clean&simple&stable" solution for incoming call popup (in browser)? i'm using flash operator panel now but i want something without flash (maybe something in AJAX?) thanks --------------------------------------- Marek Cervenka =======================================

Is there a way to tell if an extension is in use? We run a call center and it would be helpful for the people taking calls to see if we are on the phone or DND. Is that setup in Asterisk or on the phone? the phone as busy lamp field but i will just turn on after a while even if the extension is not i use. the FOP in FreePBX doesn't appear to be that helpful. i am not sure what it is supposed

Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller

I'm looking for a graphical call manager for Asterisk. Ideally this would run locally on a desktop system or be web based. I'm familiar with iSymphony, FoP, HUDlite, AstAssistant, ADM, etc but I'm finding a common problem. They all let you transfer calls that connected to an 'extension' but not 'any' call on the system. For example, when a party calls into an * system, even if they aren't speaking to someone at an extension, I'd like to transf...

Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue