similar to: DISA alternative

Hi Friends, I have configured DISA. But, its not working. When I dial to my zap channel, its asking to enter pin number. After entering PIN number, its giving continuous engage sound and hangup. Can anybody send me correct working configuration for DISA? Looking forward to your response. Thank you. Regards, Chandra. --------------------------------- Sponsored Link Get an Online or Campus

Dear list, I have problems with DISA on an specific server with Asterisk 1.4.26.2. After starting DISA I can only press one key and DISA is jumping direct into the context without waiting for further digits. In dtmf.log I found this: [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin '7' received on SIP/214-00d92db0 [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin passthrough

DISA is currently disconnecting when I dial 8888 to access DISA. Below is my extensions.conf file from A@H and some lines which shows the disconnect. Should DISA be loaded as a module in modules.conf? When I do a 'show applications' i see that DISA is there. Help! -------------------------------------- ;Asterisk CLI as I placed a call from cell into the system. Playing

Hey, this is happening to anyone who I try this with. We get into the DISA, then hear the dial tone. Dial 1 then start dialing the number, and it hangs up. I thought adding a wait time after the DISA may help, I was wrong. Here is what I have thus far in the DISA extentions. [DISA] exten => 7,1,DISA(no-password||"Scheda" <565> 455-1337) exten => 7,2,Wait(45) exten =>

I'm having trouble getting an AGI exec command to spawn app_disa. The script executes properly, but does not spawn DISA. The CLI gives no helpful clues. Am I doing the exec incorrectly? I want to have a way to authenticate callers to the extension by Caller ID... if their caller ID is in my database and set to active, they can call out. [like a calling card but auth'd by CID instead

Hi, I'd like to use DISA properly for my case - I'd like to handle it right, if user when in DISA doesn't dial any number - how does Asterisk return from DISA cmd ? I'd like to dial some default number if user doesn't dial anything or give him some message - but I don't know what gets executed after DISA if nothing is dialed .... I'm reading this on wiki, but

Hi Friends, I am facing a strange problem with DISA. I have installed and configured Trixbox. I've created a secret extension i.e., 555 and called this extension in Digital Receptionist using custom extension i.e., created in extensions_custom.conf file. When I call from my mobile phone to my PSTN number, which is connected to FXO port, my IVR is responding. After entering my DISA

When I use DISA I get congestion when I try to reach 1-800-number: Here is the context: [disa] exten => 087,1,Answer exten => 087,2,DigitTimeout,8 exten => 087,3,ResponseTimeout,20 exten => 087,4,Authenticate(985) exten => 087,5,DISA(951|disa-access) [disa-access] include => tollfree include => outgoing-voipjet [tollfree] ; ; terminate toll-free no.'s via fwdnet ; US

We are using DISA with local SIP users. The user enters in a 2 digit code then they get a dialtone and the phone dials out. The problem is that the calls waits 10 seconds after the outgoing number is dialed, no matter what I put for the timeout values. Anyone else using DISA that has run into this? exten => _2X,1,Answer exten => _2X,2,DigitTimeout(2) exten =>

Hi Folks, I'm trying to test out Asterisk overall. I'm having some problems with DTMF. Currently I'm playing with DISA, but I'm worried this will happen when I get to implementing AAs etc. I have a free SIP trunk from IPKall that I'm trying to make work. I'm able to receive calls, and I've now setup and extension with DISA and a password. I connect ok from the

Hi, These days I run into something quite odd. I have an A@H that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a

Hi ! Did anyone had issues/managed to solve issues with DISA over Zap channels on * 1.0.X (STABLE) ? I have a situatuion where DTMFs that should be recognized in DISA work over SIP channels and do not work over ZAP channels (Zap channels are on TE110P) I have in default context: exten=> 299,1,DISA(no-password|default) and I have SIP extension 200 in [default] and I have Zap trunk which

I've commited changes to Voicemail2: * Handle properly when being left a message while checking VM -- this should fix the "saving to your inbox" issue too, at least in principle. And to DISA: * Properly handle extensions with multiple matches and "dots" Please let me know on or off list about any feedback you have regarding these changes. Mark

Lists, ? ? Hi, good day, i was being task to create a DISA access for internal purpose of the company, i'm having a problem to work with it with authentication, but i think it's really a straight forward thing to do, can someone enlight me on this. thanks ? sample code snippet ? ???? exten => 5,Goto(inward,s,1) ? [inward] ? ?????????? exten => s,1,Disa(1234|outgoing) ?????????? ;

Hi I am using DISA to get my Polycom SoundPoint400 with H323 firmware to connect to * I have it working, but when I dial SIP end points there is no ringing tone on the phone. DISA gives dial tone but does not give ringing (if I understand correctly it is because it expects to transmit sound created by terminating side of the call) Is there a way to make DISA application to generate ringing

Hello all, OS OpenSuSE 10.3 * ver 1.4.26.2 zaptel ver. 1.12 Digium TE122 I have a request for remote users to be able to dial through the system so that the sales managers can barge/chanspy on the sales force. I have the DISA part working with authentication(rather straight forward) but what I can not figure out is how to enable the supervisors to be able to barge on these calls. Is there a

Hi, I have just setup a DISA setup whereby people can dial in, authenticate, are given a dialtone and can then call out. Everything works however there is a 10 second delay after the user enters the number and presses #, until the system does anything. Here is the relevant section from my extensions.conf: -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi all the help output for DISA ends like below, with the half-sentence 'Note that in the case' what's the rest of that sentence? The file that contains the passcodes (if used) allows specification of either just a passcode (defaulting to the "disa" context, or passcode|context on each line of the file. The file may contain blank lines, or comments starting with

When using a SIP phone with Asterisk, hitting the # key (pound or hash depending on where in the world you happen to be) tells Asterisk that there are no more digits coming, and to put the call through immediately based on the digits already entered. This is the same functionality as the PSTN (at least in North America). However, DISA just sees the # as another digit, and therefore pressing #

Hi/ I am newbe in asterisk. I try to setup callback with Disa on my home server Anybody help me, pls