similar to: IVR Tree Best Practices

OK. I'm getting out the fireproof suit because it's coming and my hackles have been raised by a number of comments on the list of late. Disclaimer: No disrespect intended to the individuals of any *specific* thread. I'm a little frustrated over energy wasted on pedantic top/bottom posting crap rather than understanding the technology and industry best-practices which have been

Hi, We?re going to build an IVR system with a TE405P and 4 E1. We?re sure that the 120 channels will be filled by 120 simultaneous calls during peak, so we want to have the good server to manage this. We wonder a lot of things and maybe you could help us. - Are you ever build a similar system ? - What hardware is the most adapted for this type of use ? - What is the most important ?

Hi, I have a couple of users who are having a peculiar problem. On some outbound numbers where there is a deep IVR tree (3+ selections), and then a live person picks up, the live person will be unable to hear them on the phone, but they can hear the live person. I've done packet traces and it appears as though audio is being passed both ways, but the audio from the caller is severely muted

Does anyone know why? ThePBX*CLI> -- Executing [310-456-7890 at from-trunk:1] Set("SIP/202.101.202.101-b763ce60", "__FROM_DID=310-456-7890") in new stack -- Executing [310-456-7890 at from-trunk:2] ExecIf("SIP/202.101.202.101-b763ce60", "1 |Set|CALLERID(name)=310-456-0987") in new stack -- Executing [310-456-7890 at from-trunk:3]

hi all, i have gradwell DID i am using it for inbound dialing with IVR when ever customer call my DID some times other IVR is cumming on my IVR that IVR is not even related with my server .can u please help me on this -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have setup 2 trunks using 2 different voip providers using sip. the first one i have no problem calling inbound then redirected to an IVR, i can hear the IVR. the second one has issues, inbound works going to IVR as i can see it on the CLI, but i don't hear anything. i tried redirecting it to an extension not an IVR just to see if inbound really works, and it rings the

We are running 1.2.14 with an IVR in the dialplan. If I connect to the IVR with a SIP phone (Polycom or Xlite) and press a couple of digits very rapidly (I found this with 33 on a sticky keypad) which are an invalid response, Allison will go into a loop saying 'I'm sorry, that is an invalid response, please try again.' over and over. This does not happen with a commercial

Dear all is it possible to handle a queue using a programmed IVR? As i understood, is possible to: - do some IVR in the dialplan BEFORE to queue the call - put a timeout to exit from the call and then do some IVR in the dialplan - intercept a single dialtone to exit the queue and performe some IVR in the dialplan (context setting in the queue) I've tested these things but in each case if i

I am testing an ivr but I'm having problems. The call keeps looping and it doesn't hangup the call after passing three times through the menu. Here's my conf: exten => s,n,NoOp("Here's Count") exten => s,n,NoOp(${COUNT}) ;123,n,Set(COUNT=$[${COUNT} - 1]) exten => s,n,GotoIf($[${COUNT} = 4]?33,1:44,1 ) exten => 1,1,goto(tech-support,s,1) exten =>

OK, maybe not even best practices, but at least, tested practices. >From my reading thus far, it is evident I shouldn''t use Rails to handle mongrel log rotation. Fine, I''m sold on that. The advice I''m seeing says that that one should use an external script to do this. So my question is, what are people using? What''s working well? Simple bash scripts or

Hey, Is there any way to play IVR and Read DTMF during active call. Call Flow: ? ?? USER1(initiator)? <-----> Asterisk1 <-----> Asterisk2 <-----> ?? USER2 How I can Play IVR and Read DTMF from USER1 When both users are in active session. I am able to play IVR and Read DTMF from USER2, which is not required, When Asterisk2 Receives call from Asterisk1, it simply

Hi, We developed a somewhat complex Asterisk-based IVR for a client of ours. When I say 'somewhat complex' I mean that this IVR is highly integrated with a set of middle-ware applications that allow the caller to perform extensive functions while navigating the IVR. This has been working great for the past year or so. However, now the client wants to add the ability for callers to be

Dear All. I'm very new to CT, but attracted by asterisk. I plan to start learn to build IVR, based on asterisk. Do I need FXO/FXS card to start building simple IVR box ? Can I use OEM X100P - FXO PCI Card ( http://www.digitnetworks.com/store/product_info.php?products_id=28 ) to start build an IVR box ? What is the cheapest FXO PCI card that can use with asterisk to build my first IVR box ?

Hello Team, IVR selection of QUEUEMETRICS As we know queuemetrics had an IVR selection functionality where it can get the IVR keypress of a caller. We saw this link http://forum.queuemetrics.com/index.php?action=printpage;topic=503.0 and upon checking, its only determined the Queue, I want to get is the per IVR of a caller. Can you help me guys regarding this? I want to implement this with

Hi, As Freelance programmer/consultant I'm looking for project/task of IVR/CTI/CRM/IP-based, my skils are as following, 1 Dialogic-based CTI/IVR software programming 2 Intervoice IVR development 3 Siebel CRM integration and development 4 IBM DirectTalk and WebShpere Voice(VoiceXML,...) 5 IP-based development(VoIP,h323,sip,...) 6 Cisco

Hi All, I am using Asterisk 15 server and wanted to configure IVR call simulation. My configuration scenario is 1. A subscriber will register to Asterisk server and start a call. 2. The IVR audio will come from the Asterisk sever to sbscriber. 3. Once the subscriber pressed the botton, the call will connect to a number based on DTMF digit pressed by subscriber. Then call will continue for 30

Is anyone else experiencing problems with Asterisk 1.2, the ivr does not work. I have tried it on Linksys RT31P2 and Grandstream Handytone 496. After a call goes through you're not able to enter any of the prompts on a IVR. and cannot enter pin numbers when using a calling card or anything that requires you to enter into an ivr system. I already set my dtmf mode in asterisk. --------------

Hi there, I have the following scenario: I have a GK that controls my H323 GWs. And, I am needing an IVR royalty-free system. More than an IVR, it's only something that should play a sound file depending on the called digits. So, I wanted to use asterisk as my IVR. So, I've installed asterisk with the chan_h323 module and used the Playback comand in my extensions.conf

Hi list i setup successfull asterisk version 1.4 + opensips, Opensips is the Registrar Server, Asterisk is the IVR server the call flow IP phone ---INVITE 1001----> opensips -----> ASterisk ----INVITE 5001--->opensips ---> Busy|cancel|404..--->asterisk---wait 10s to bye --->IP phone (5000) my case is: 1/ IP phone(5000) --->Opensips 2/ IVR number : 1001 3/ IP

I'd recommend using DEVICE_STATE On your extension 101, Check the DEVICE_STATE of peer SIP/101, If it's not 'NOT_INUSE' then dial it, Otherwise dial SIP/102 exten => 101,1,ExecIf($["${DEVICE_STATE(SIP/101)}"="NOT_INUSE"]?Dial(SIP/101,40)) same => n,Dial(SIP/102,40,t) same => n,Hangup() On Wed, Feb 25, 2015 at 2:08 PM, ricky gutierrez