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  • Faxing through Trixbox/Asterisk.

    We have a Trixbox 2.8 phone system using a SIP trunk that we would like to be able to easily fax through. Currently, I have HylaFax set up to send faxes out using the Frog Fax client on WIndows 7. This works fine, but the steps are rather clunky to get other users to fax. First, we have to scan a document...
    Posted to Asterisk (Forum) by network on 03-14-2010

  • Asterisk... silently listen to DTMF tones.

    I have a asterisk server whick works in the following way. When asterisk recieves an incoming call, it creates a conference room using 'meet me' and puts the caller into the conference room. Later other people maybe added to this conference room. My problem is the following, I want the caller...
    Posted to IP PBX (Forum) by network on 03-13-2010

  • Polycom 330 - no ringtone on forwarded calls

    Hello all, I'm using a Polycom SoundPoint IP 330, and when I use the built in call forwarding feature, incoming calls are forwarded properly, however, no ringtone is heard by the caller. Is it possible to enable a ringtone for forwarded calls? Thank you.
    Posted to IP Telephony (Forum) by network on 03-01-2010

  • Asterisk Dial Plan works great but I need to place caller in meetme then kick off another call. see code.

    I have a dialplan where I need to put caller in meetme that create dynamically (this works fine) Then I need to call another number and put them in that same meetme. It seems to stop the dial plan once it goes in Meetme. Is there a way around this without AGI script. Using dialplan. DialPlan::::::: ;;Vars...
    Posted to Asterisk (Forum) by network on 02-28-2010

  • How To Add Multiple Caller ID Numbers To Inbound Route in TrixBox CE

    In TrixBox CE, attempting to add multiple numbers in the field labeled Caller ID Number under Inbound Routes. Currently have an extension setup for a voicemail only mailbox. Would like to access this mailbox from an external phone number but want to limit access to this box to two ( 2 ) particular cell...
    Posted to Asterisk (Forum) by network on 02-24-2010

  • how to send call to exten using GotoIfTime  application

    i am having Dial plan in which call comes and it goes to ext 112, no problem in that. WHAT I WANT 1 - Office close at evening 8'oclock i.e 20:00 and opens at 8 am. now i want during this close time the call should be attended by security person in his ext is 113, 2 - if he dosent attend call in 30...
    Posted to Asterisk (Forum) by network on 02-22-2010

  • Transfer (T) in Asterisk

    i want to use Transfer Function (T) i.e when ever caller calls and if his call is not answered he should hear the voice, enter the extention of the person.n if he presses the wrong extention he should hear u had entered wrong extention.if he dosent dial in a given period of time than it should time out...
    Posted to Asterisk (Forum) by network on 02-22-2010

  • How to Configure Channels in Asterisk

    i had installed Sangoma USB Fxo. now machine is detecting FXO. i have 2 FXO slots in Sangoma which i have to configure in asterisk.below is the Sangoma Detail [root@localhost wanpipe-3.6.0.12]# wanrouter start Error: Device wanpipe1 is already running Run 'wanrouter stop wanpipe1' first Configuring...
    Posted to Asterisk (Forum) by network on 02-21-2010

  • how to tranfer call on mobile from asterisk

    hi i have configured asterisk in which when calls come in PSTN exten 112 vi extensions.conf [from-zaptel] exten => s,1,Wait(2) exten => s,n,Dial(SIP/112) [my-phones] exten => 112,1,Dial(SIP/112) exten => 113,1,Dial(SIP/113) exten => 114,1,Dial(SIP/114) what files to edit so that when call...
    Posted to Asterisk (Forum) by network on 02-21-2010

  • DTMF issues in Asterisk

    I am trying to setup asterisk with A2billing to provide callback service. The problem is DTMF do not pass most of the time. I tried different carriers, all of the carriers have this issue. Is it truely a carrier related issue or Asterisk Issue?
    Posted to Asterisk (Forum) by network on 02-21-2010
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