similar to: Call waiting / hook flash on ZAP trunk from SIP phone?

Problem resolved. This makes it nice and simple to 'flash' an incoming POTS line (ZAP channel) as opposed to the dialplan scripts that I have seen that require tranferring the call, hanging up, and waiting for a call back. That was too confusing for my wife. Now all she has to do is pres *3 and it is done. No transfers. No hanging up. No dial back. extensions.conf [context] exten =>

Hi Guys!! I wrote a little patch for asterisk 1.2.5 and I will maintain it for future release unless somebody explains me how we can ask people at Digium to add it to the source tree... We are planning on using Asterisk as our main PBX for the office over the next few weeks. Our current setup uses TDM400 cards to bring our 8 lines into Asterisk, our Telco provides us an option for three

Hello, Is it possible to detect a hook flash in asterisk. I want to be able to perform some functions an hook flash. I have the following entry in features.conf which executes a Macro on detecting key press '**'. [applicationmap] test => **,caller,Macro,testflash Is it possible to do this action on hook flash? -------------- next part -------------- An HTML attachment was

Hi, I hope someone could help me, i have a x100p interface for testing purpose and on each incomming call I redirect the call to handytone 388 atas, the problem comes when i'm during a call and another call comes in, i hear the call waiting beep (comming from the zap channel), but I can't catch the call as usually using flash+2 (my pstn call wait sequence), because when i flash the

Hi everyone, our setup includes a NEC PBX connected to our asterisk via bri lines. The NEC has a doorphone feature, which is just an extension that calls you when someone rings. When connected to this extensions, a 'flash' signalling opens the door. So now, i'd like to trigger this from asterisk, too, but unfortunately wasn't able to do so. Setup: asterisk

In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here.... Any help is appreciated. Here is features.conf: ; ; Sample Parking configuration ; [general] parkext => 700 ; What extension to dial to park parkpos => 701-720 ;

Hello. I'm trying to install Exchange 2013 (SP1 CU5) on a domain with only a Samba ADC (because I've had many problems trying to get OpenChange to work with Outlook 2013.) I've already hacked the attributes of the ADC's LDAP entry so Exchange is happy with the OS version (see my previous post,) but I want to find out if there are any known problems with letting Exchange make

hi all i have samba4 ad setup and working, i am currently trying to set up passwordless ssh on my client servers, i have read this page https://wiki.samba.org/index.php/Authenticating_other_services_against_AD i have a properly configured krb5.conf file, i have a keytab from the samba dc and i can kinit and obtain a valid ticket. the only thing i have not done is to join my client which is a

Hello... I have a big problem with asterisk. Every time i make a call asterisk does not bridge the zap channels. The zap channel from which i'm calling remains in state:ring and applicaton:dial and the zap channel with the external line configured remains in state:dialling an Application:AppDial. Zap/20-1 agentie s 1 Dialing AppDial (Outgoing Line) 09399 (None) Zap/9-1 int_omg 09399 5 Ring

Hi there, I have a x100p card in an asterisk box. Does anyone know if it's possible to do a hook flash / recall on an active zap channel? On what I'm trying to do... >From an ordinary analogue pstn telephone I can call someone, press recall, call someone else, press recall 3 & presto we're on a three way chat, with me only using one line - using the telephone company's

I just registered ID 0004283 at http://bugs.digium.com for the problem described in subject (found when using a Linksys PAP2-NA). I don't know where the proper forum is to discuss, so I'm hoping anyone interested will read the bug and let me know your thoughts, either at bugs.digium.com, here, or by emailing me directly (or, please suggest another forum that is more appropriate). As

Do you have threewaycalling and transfer set in your zapata.conf? Here's mine (four TDM400's, seems to be working so far). I didn't do anything in my extensions.conf for any of these features (what confused me at first is the t and T options of the Dial application in extensions.conf are for transfers via the # key), when you flash you get another dialtone that works just like the

I have been looking for the answer to this question for a while. Google-ing and reading the archives of Asterisk-Users has not enlightened me. It seems that this question has been asked many times, and many times it has gone unanswered. I have call waiting and three way calling on my PSTN line from Verizon (the local telco). This is connected to a FXO port on a TDM400P. I also have

Hello. I just ran into this issue myself (with Exchange 2013) and found out how to solve it, so I'm trying to reply to this thread to help anyone else who is looking. Use an LDAP editor (I use Apache Directory Studio) to edit the entry for the computer account of your (master) Samba domain controller. It has a DN of the form: CN=<computer name>,OU=Domain

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I was trying to update the group policy with gpupdate /force and was getting an error message saying can't connect to \\example.local\sysvol\example.local\Polices...etc So when I tried to navigate there I could not see the Polices folder. I ran samba-tool ntacl sysvolreset and tried again, rebooted...still nothing. Then instead of using example.local, I just used the IP, 192.168.0.3 and then

Hello, I have problems using dynamic features while an other person is doing call waiting in a call. I define a dynamic application mapping in features.conf as the following: testfeature => *9,caller,Playback,tt-monkeys I also set DYNAMIC_FEATURES => testfeature. The mapping is working well. But during a third person is calling I'm hearing just the call waiting tone and none of my

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, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user