similar to: Hangup?

Try using two IVRs. The 1st 'Intro' with your 'will be recorded' message has a 1 second timeout, and the only entries in it are 'i' invalid (points back to itself), and 't' timeout (points to the 2nd IVR for dialing). > Date: Tue, 25 Apr 2006 21:25:53 -0600 > From: "Carl Youngblood" <carl@youngbloods.org> > Subject: [Asterisk-Users] Trying

I have these 3 models. class Ivr < ActiveRecord::Base has_many :klusterings, :dependent => :destroy has_many :klusters, :through => :klusterings, :uniq => true end class Kluster < ActiveRecord::Base has_many :klusterings, :dependent => :destroy has_many :ivrs, :through => :klusterings, :uniq => true end class Klustering < ActiveRecord::Base belongs_to :kluster

hi, all I have problem to sense digit in my ivrs. scenario is below: I am using zaptel T410P digium card to competible with my PSTN(CORAL) [ivrs] exten => s,1,Background(welcome-ivrs) exten => 1,1,Playback(welcome) exten => 2,1,Playback(goodby) sound file are .wav files. when i dial no. from analog phone to launch ivrs welcome-ivrs.wav file plays and when i press digit 1 play wecome

Hello people, this is my first mail to this list. I'm new to asterisk and trying to set up an IVR. So far my dialplan works nice connecting with softphones, but I'm having problems to detect hangups on the analog line. Here are the details: -Clone pc with Ubuntu server 10.04 64 bits -Asterisk 1.6 -Openvox A400P, lspci -v says: 01:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX

We have a asterisk installation we are using for a hosted PBX solution.. we chose to use 10 digit extensions... We separated our customers by contexts and have encountered a problem where one customer can't call another using 7 digits.. even if we prepend the area code when 7 digits are dialed... anyone consider reviewing it and making recommendations? We're reluctant to post

How are we all documenting complex dialplan? Is there something similar to Doxygen? I've got around 20 config files covering around 60 contexts and 40 variables. Of course, I've maintained a basic list of the major stuff, and documented the code throughout, but it's grown to the stage where it needs to be better documented, have a proper flowchart etc. Talking of flowcharts, I see

Hello, since applying the patches related to XSA-36 Xen recognizes a broken IVRS ACPI table and disables I/O virtualisation. I contacted the manufacturer of the mainboard/BIOS and they want to help me by providing a patched BIOS - so far so good. However, they need details about what to fix, which I don''t know either. Could you pls. give me some hints which I can forward to the

I thought I had posted this, but I didn't see it in the archives, so I guess I hadn't. I've got FXS lines going to a legacy IVR. When I Dial into one of these lines and then hang up, FXS plays the Congestion tone until the IVR drops voltage. I would like the IVR to hang up sooner. I could do this by either making the IVR recognize the standard Congestion tone, or changing the

sorry, was only for users list... Hi Kevin, Hi list, you are right, acting now is not needed, when callbacklogin will be removed anywhere in future... But thinking how to realice alternatives can't be so wrong. Callbacklogin gives a very simple way to use more queues for one agent, which only has to logon to only one system. No need to make dbs or tables for saving, where the agent has to be

Good day , I have configured TDM410P (asterisk 1.6.x) on Cent OS 5, but dahdi take some time to hangup the call when playing the IVR..(it will send the hangup signal after finishing the IVR promt..) is there any specific setting to avoid such incidents ? iam using busycount as 3, signalling=fxs_ks ;toneduration=100 callwaiting=yes threewaycalling=yes callreturn=yes

Hi, We are running 1.4.22 and have been experiencing problems with certain IVRs and DTMF Tone duration. We would like to be able to increase DTMF Tone duration by 50 to 100ms over what the user is pressing on his phone. We have a PRI test circuit and an analyer in between to measure tone duration. We have tried setting chan_dahdi.conf parameter 'toneduration', but that does not do

I don't know if this is a problem with my cisco sip IP Phones or asterisk but I thought I would post here in case someone else has experienced this issue. When I make a call from my SIP cisco IP Phone to some remote IVRs I never get the rest of my soft keys, only the "End Call" soft key, and also DTMF doesn't work... its like the phone is acting like the remote end hasn't

From: Suravee Suthikulpanit <suravee.suthikulpanit@amd.com> This patch handle additional cases for IVRS bugs where special->handle is not correctly initialized for IOAPIC and HPETS due to firmware bugs. Signed-off-by: Suravee Suthikulpanit <suravee.suthikulpanit@amd.com> Provide logic in "is_ioapic_overidden()" Signed-off-by: Jan Beulich <JBeulich@suse.com> ---

Hi everybody, I'm running an IVR menu with different languages setted up by user when they enter this menu. What I want is when they hangup, asterisk sets the default language (aka en) back. I'm wondering which extension is called after a hangup in a background command? BTW my IVR menu is in a goto context. -- Daniel

Hi, I discovered yesterday that the instructions given at http://www.openssh.org/faq.html#3.14 regarding the correct permissions for the authorized_keys file mistakenly recommend chmod'ing the file to 600 when it should be 644. The requirement for public key authentication to work is in fact that ~/.ssh/authorized_keys is readable (but not writable) by group and other, not just owner. Someone

Doug, This is where the weeds start growing. On 6/17/2023 4:55 AM, Doug Lytle wrote: > > For both capabilities, you can use Background() instead of Playback() > for audio prompts.  Background() allows for interrupting the prompts > and continue on with your dialplan. > > Understood. From the book: The most common use of the Background() application is to create basic

Hi All; Why busydetect=yes caused this autmatic hangup happens without any reson (while responding to entering the digits in the IVR) I do not know! And what is the solution I do not know. I used busydetect=yes and busycount=5 in zapata.conf to help me in hangup when detect the busy tone, but I faced the following problem: While I am calling to the Asterisk PBX, and during the response for the

I have a Vonage acct that uses the Cisco ATA186. Currently, I have the ATA186 plugged into a SPA3000 to act as the FXO port. I installed a X100P card with the idea of replacing the SPA3000. Now, when I plug in the ATA186 into the X100P card and make a call into the system (from cell phone) and hangup when the IVR is playing, Asterisk is not detecting a hangup and keeps looping the IVR. If

I upgraded from 1.0.9 to 1.2.1 My IVR which worked perfectly on 1.0.9, now hangup with no reason (at least I could not find a cause) When this hangup happen, I can read: == Auto fallthrough, channel 'IAX/user-20' status is 'BUSY' This happening also with ZAP channels I'm really disappointed with 1.2.1, what is benefit from upgrade if I must spend couple days to get my system

Hi All, I'm using Asterisk 1.8.2 with outbound calls using Google Voice. I've been having issues calling several toll free numbers where the call 'is ringing' but never transitions to 'answered'. These are toll free numbers which are typically answered by an ivrs where you enter eg. a conference bridge number. I searched google and the closest reported issues I found are