similar to: Attn: DISA Experts(Strange problem with DISA)

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

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

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

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone

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 =>

Here is some code to do authentication by caller ID for DISA through AGI. My original code had a bug in the Mysql query code, and there was a hangup in the wrong place [that's what I get for coding something at 2:00am], but the attached code works correctly. Take note of the REGEXP for the CallerID variable. When I tested the code from the PSTN it worked because there was no name component,

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

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) ?????????? ;

Jeez, I need to work out the shortcut to send an email which I keep pressing by accident!! -----Original Message----- From: C. Tomlinson [mailto:asterisk_list@burntwires.com] Sent: 27 February 2005 22:48 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] DISA and a long delay; ideas? Many thanks, that was the problem. I didn't paste the

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 =>

Before I go nuts trying to figure this out, is anyone using DISA in this manner? exten => s,1,DISA(XXXXX|context|callerid) Everything works except the caller ID part. What I had wanted to do is to setup up a file of authorization codes where each code was associated with a context and caller id. The format for the file, I thought would be: XXXX|context|callerid and I had it set up in

Hi, I'm trying to configure Asterisk and DISA. Asterisk is working, but I cannot have DISA dialing out. This is a snippet of my extensions.conf: [internal] exten => 1003,1,DISA(no-password|outgoing2) [outgoing2] exten => 1003,1,Playback(beep.gsm) exten => 1005,1,Playback(beep.gsm) My understanding is that if I dial the extension 1003, I should then be redirected to the context

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

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

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 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

Hello, Im tryng to setup DISA on my server. Outlines comes via VoIP to my asterisk server. When i dial from outside to my disa number it answers. I dial the extension that i want to dial but Dial tone keeps up playing about 3-4 seconds more even i start to enter numbers..Then when timeout occurs, it dials the number. What is strange is even if detects the dialed number in a mysterious way, if

[moved from -dev, as the thread is better suited for -users] At 5:10 PM -0600 1/30/04, James Sharp wrote: > > I've pretty much got the routing covered at this point, I'm just not sure >> how to get the Asterisk system to answer and give me dialtone immediately. >> Any ideas or recommendations would be greatly appreciated. > >app_disa will give answer and give

'http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA' This should help. -----Original Message----- From: Michael Graves [mailto:mgraves@mstvp.com] Sent: Wednesday, January 26, 2005 9:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] dialplan logic for conditional DISA on incomming 800 number Hi All, I have an 800 number from

I have a setup which allows users to access my asterisk box via FWD. That is, a user in say, France can call into a local access number for FWD, then hit number 7 which dumps them into a DISA request for a password, which then dumps them into my internal extension so they can dial out through a VOIP provider. Everything works fine until they enter their tones for the number they are calling