similar to: disa and caller id

I'm getting Number but when I look at the CDR database. I do see the name -----Original Message----- From: Lacy Moore - Aspendora [mailto:aspendora@gmail.com] Sent: Mon May 01 17:10:26 2006 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] CallerID Name problem Do you get caller ID number? If so, WAITing is not going to help, since you

The "telco guys" probably did something non-industry standard and reversed send and receive in the jack that they plugged the CAT5 into. Sure it works, sure it is easier, sure it is not the correct way of doing things. Thanks, Steve ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Lacy Moore - Aspendora Sent: Sat 4/22/2006 2:55 PM To: Paul

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,

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

On Mon, 2007-02-05 at 12:00 -0700, asterisk-users-request@lists.digium.com wrote: > Date: Mon, 5 Feb 2007 11:36:28 -0500 > From: Andy Davidson <andy@nosignal.org> > Subject: Re: [asterisk-users] Question on G.729 (was: H.264 *Not > Patented*) > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >

I'm wondering if anyone else has experienced this. Up until a few days ago, when accessing the CLI from my terminal program (Private Shell), the output was in color. I haven't upgraded, rebuilt, or to my knowledge, changed anything in Asterisk that would change this. My terminal settings were the same as well. I have two computers that I access the CLI regularly on, and neither show

I just noticed the Aastra 57i do something that I haven't seen before. I called from one phone (phone 1) to the 57i. I answered it. Then, I pressed Transfer and dialed the extension for the third phone (in this case a Cisco 7960 in Sip). I did not answer the Cisco, but noticed the caller ID was showing the Aastra (as expected). I hung up the Aastra to complete the transfer and noticed the

I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. It does work when putting the call on hold. If I revert back to 1.2.23 using the same config and same music on hold files, it works. I've looked at the sample config files for 1.4 and nothing seems to jump out at me as to what

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

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 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 =>

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

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 =>

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