search for: parkcalls

Hello all, I want to park calls using the callpark application, but I don't want to hear the saydigit when the called is parked. To resolve this issue I use the following instruction in the dialplan: exten => _8XX,1,ParkAndAnnounce(|1000|local/10 at default|) Because local/10 at default is not defined to a peer I get a lot of warnings. :( Is there a better way to resolve this

Hi folks, I was using the following featuremap: blindxfer => *1 disconnect => *9 atxfer => *2 parkcall => *7 automixmon => *0 and everything worked. Then the need arouse to use some features like automixmon during a conference, but MeetMet has the * key bound to the (admin) menu. Thus, in order to enable features like automon and transfers even during a conference, I

My company for various reasons has asked that I come up with a way to have previously parked calls be re-parked in the same parking slot. I have looked at setting up asterisk so that the receptionist chooses which slot to place a call, but I think there is an easier way. That is when I came up with the idea of "Sticky Park". Here is how it would work. A call would come in and

Hello asterisk-users, I setup my asterisk to support several features like automon,blindxfer,atxfer,parkcall etc. by using features.conf and the global variable DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in extension.conf. Every Dial() command in my diaplan has the appropriate parameters out of {tTkWwW}. For calls from my SIP phones everything works fine. Pressing #1 will

Hi, How can I park a call from dialplan and get going?? Example: 1. Answer 2. While follow = false 3. ParkCall 4. Checksomthing ? follow = true 5. Endwhile 6. UnParkCall 7. Go on .. The idea is let the call waiting while I do some things on the dialplan, is it possible?? Maybe is not parking the solution?? Thanks

Greetings all, I'm trying to connect to an AT&T teleconference, but the call is never marked as ANSWERED by asterisk and therefore won't bridge and continue. The only work-around I've come up with so far is to dial like this: Exten => 744,1,Dial(Zap/g1,,p) The "private" mode keeps the line open without trying to do a bridge, but requires the

Hi, Been playing with Call parking, and I can`t help but wonder if I am doing something incorrectly. The way I understand it (using default config in features.conf), is I would transfer a call to extension 700, which would park the call, tell me "701". I could then hang up, go fetch the fright person and tell him "call 701 you have a call waiting for you". The way I

Hi everybody, I would like to set asterisk up such that to use the call parking feature, instead of transferring a call to the extension set up in features.conf, you just dial a code (e.g. *3) and this then parks the call. The main reason for this is that a number of the phones I use have transfer buttons that I can't reprogram to use Asterisk's own transfer functions, therefore you

I am trying to use call parking. I have the following in features.conf [general] parkext => 700 parkpos => 701-720 context => parkedcalls When I try #700 from my softphone asterisk just passes it and doesn't interpret it. Can someone tell me what I am missing? I am using asterisk-1.2.17 Thanks, Alberto

like this: print STDOUT "EXEC background /var/lib/asterisk/sounds/wait-moment \n" system("program2.agi &") exit; program 2 would run while the sound played. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric "ManxPower" Wieling Sent: Tuesday, December 02, 2008 3:20 PM

like this: print STDOUT "EXEC background /var/lib/asterisk/sounds/wait-moment \n" system("program2.agi &") exit; program 2 would run while the sound played. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric "ManxPower" Wieling Sent: Tuesday, December 02, 2008 3:20 PM

One more Polycom IP601 question please (sorry for the long intro here to document) ... In order to closely approximate the behavior of the previous telephone system that many of the users are familiar with, I have set up call parking like this: - features.conf [general] section contains: parkext => ** ; What extension to dial to park parkpos => 10-11 ; What

I configured "parkcalls" and can see on "cli" Playing "digits/7" etc but I cannot hear them in the phone. I have: features.conf. [general] parkext => 700 parkpos => 701-720 extensions.conf: [extensions] include => parkedcalls What did I miss? -- #Joseph

i've got the parking lot set up in asterisk 1.6.2.8. when a caller calls into the pbx and connects to an extension, the answering extension can place the person into the parking lot by dialing #72 during the call via the parkcall parameter in features.conf. this works just fine. however, when another extension picks up the call from the parking lot (e.g, by dialing x701), that extension

Just upgraded to 1.8, we use the multi lot parking feature by dialling *4. We are not getting the parking slot announcement being played to the person who parks the call, so it's impossible to tell which slot they've gone into. Could someone check our config? On Debian Squeeze using packages from http://packages.asterisk.org/debsqueeze main (Asterisk 1.8.11.1-1digium1~squeeze)

I have a SIP phone calling via a SIP trunk another asterisk system, that then sends the call out a ZAP channel. When I press any of the features defined in features.conf, The end user on the ZAP side hears the DTMF tones, and none of the features work. My DTMFmode on the SIP users definition is rfc2833 Asterisk console doesn't register that a feature is being recognized, any ideas? Below

Dear all I have beginer in Voip and i have configured Asterisk server with 100 IP SIP phone ( SNOM ) everything is fine but problem is how to transfer call from one user to other means i call to some one and then someone want to transfer call to another person how it is possible i have also try with feartue.conf but it is now working i have also read document on voip-info website

The Asterisk Development Team has announced the release of Asterisk 13.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.9.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Certified Asterisk 13.13-cert1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 13.13-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the

The Asterisk Development Team has announced the first beta of Asterisk 14.0.0. This beta is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.0.0-beta1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this beta: New