similar to: Using DISA when dialing into a Zap interface

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

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, 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, I'm getting unusable DTMF detection with DISA on incoming ZAP channel (bristuffed *) on quadbri from GSM gateway. DTMF detection works ok in normal ISDN incoming line. How can I check what's going on ? What settings to check ? Anyone with more experience on such scenarios ? Thanks in advance, regards, Rob.

Hi guys, I am trying to use DISA. The scenario is - I call my home number (where X100P seats) from mobile phone, enter the password, enter international number and get connected via voiptel. It works perfectly when I call extension setup with DISA from X-PRO SIP phone, but when I dial into Zap, It seems that it does not detect DTMF tones. Here is a log and config files Please help

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 tried to google this issue with no resolution. I'm having the same issue as this person: http://lists.digium.com/pipermail/asterisk-users/2004-August/058280.html Basically, anytime I try to dial out on my TDM400P w/ FXO, I get "we're sorry, but your call cannot be completed as dialed." When I "debug channel Zap/x-x", I get a whole bunch of this: [ TYPE:

I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 & 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure

I have one POTS line going into a TDM400P. Here in Atlanta, we have 10 digit local dialing. I launch a call "Zap/1/7705551212" and it goes thru just fine. The next time I try it, without any modifications, I get a Bell recording telling me that I must dial the area code and seven digit number when placing a local call. It's like Asterisk may be starting the dial before the line

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 =>

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

Hello all. This is my very first message to the list. I have a TDM400P card, It has 2 FXO channels which are connected to extensions of my PBX (Ericsson BP250), so I can dial from any SIP softphone directly to physical (analog and digital) extensions on my company. My PBX is configured so when I dial 8 on any extension, it will redirect to the first free FXO channel on my TDM400P card.

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, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>

Hi everybody I am having problem detecting fax with my X100P. I have RedHat 8 as OS and an X100P and a TDM400P. The X100P being plugged into PSTN. I have successfully installed tiff-v3.5.7 and spandsp-0.0.1 and also patched Asterisk wthout problem. Here is my zapata.conf file context=cda signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5

....I THINK. When dialing 1+10 digits, I occasionally get a telco message "You must first dial a 1....". When I look at the console, the number is being sent to the ZAP channel properly. We're talking about a couple of POTS lines on a TDM400P. I'm thinking that it may be starting the dial too early after coming off-hook because I can just redial and have it work (or not)

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

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,