similar to: Asterisk Guru help needed for DISA troubles

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

I can dial from an asterisk host to another one via FreeWorldDialup, on the other side DISA service answer to me and i can ear dialtone. But i cannot send DTMF and dial an extension on the DISA enabled asterisk.....i've tried rfc2833 and inband...but nothing....any tips ??? Thanks, -- Igor Barsanti GPG Public key available at http://pgp.mit.edu

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

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

911 Help! I accidentially deleted all directories under /var/spool/asterisk I did use the backup facility not too long ago but cannot find the process for restore. However, I don't believe a full restore is needed -- I just need to know the names of the directories under /var/spool/asterisk and re-create them (I hope!). Can some kind soul give me some direction or tell me the directory

Hi Guys, I have noticed a weird behavior in 1.4.12. When using Authenticate or DISA in the dial plan the channel immediately switches to gsm format (if you request a password) or slim (if you run DISA without password). The debug log says... =============================== [Oct 14 21:23:00] DEBUG[9013] channel.c: Set channel SIP/1970xxxxxx-0821aad0 to write format gsm [Oct 14 21:23:00]

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

Hello all, When I call the Broadvoice number all is good. When I try to call out through DISA on my broadvoice line i get the following: Executing Dial("SIP/147.135.0.129-0815bc60", "SIP/16037862111@proxy.bos.broadvoice.com|30") in new stack -- Called 16037862111@proxy.bos.broadvoice.com -- Got SIP response 480 "Temporarily Not Available" back from

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

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

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 =>

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,

Greetings Asterisk users! I'm trying to setup Asterisk system to act as a callback system together with callcentric (http://callcentric.com) but it appears that I hit common DTMF issue and I want to workaround this problem. Basically my current setup is the following: 1) I have dedicated Asterisk server that it is linked to my callcentric account 2) I have US phone number (DID) from

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

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

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

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 =>