search for: kktt

Hi, When a VOIP user dials an external number, the calls are routed through our SIP provider. Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider? Something like GotoIfExists(5551234 at incoming_calls) Currently, I'm paying for all calls, regardless of whether they exist locally. All DDIs exist in the incoming_calls context. Thanks Dan

I have a Asterisk PBX System with Redhat Linux Fedora 4, Webmin version 1.400 and I am simply trying to configure into the "Extensions.conf" script an entry that will add to the "Auto-Attendant" a line that will allow a "Caller" to enter a "0" (Zero) will then ring the extension(s) of the "Operator" to speak directly with the "OPERATOR"

Hi! I have the following weird problem: phones A,B and C are in the same callgroup/pickupgroup. A call B, B is ringing, C calls *8. Now, B is CANCELed, C gets 200 OK, but A is still in Ringing. Is there anything else I have to configure? thanks Klaus

Hello Folks; I appreciate all of the help so far - thanks. Another question: I am using MixMonitor() to record calls and I would like to include the called number/extension in the filename: In my dialplan, I am able to save the file with the caller id in the filename. However, what I am a little unsure about is the incoming number/called number/extension - passing that information on to part

We have an employee who works from home. We sent her a SIP phone to work as an extension off our Asterisk 1.6 system, but her DSL service is so bad she was dropping calls all the time. It's not just a tuning or QoS issue. Her service is simply unreliable. She had a POTS line installed and I have the dialplan set up so that when her extension is dialed, it calls out over our SIP provider to

...Normal><font size=3D2 color=3Dnavy face=3DArial><span = style=3D'font-size: 10.0pt;font-family:Arial;color:navy'>SIP/104-08461bd0&nbsp;&nbsp;&nbsp;&n= bsp; 1-dial at macro-trunkdi Up&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = Dial(DAHDI/R1/w2975000|20|kKtT<o:p></o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span = style=3D'font-size: 10.0pt;font-family:Arial;color:navy'>2 active = channels<o:p></o:p></span></font></p> <p...

Greetings List, I have configured chan_mob for Nokia 7610. I can succefully dial from softphone to mobile and land line numbers, Softphone (PC) =====> Asterisk ====> FXO (Nokia 7610) ====> Destination Number When call is established I have to use Nokia 7610 for conversation. Is it possible to use softphone, dial via mobile phone and have conversation using softphone?