similar to: connect 3rd party IVR servers with Asterisks.

Hi? 1. When a mobile user dial an IVR short code , mobile network able to divert that call to Asterisk platform.? 2. There would be web servers which are holds Voice XML . 3. Asterisk would be able to redirect the mobile request to certain Voice XML server accordingly. Just for like this setup , how do we install asterisk with minimum of asterisk modules ? Do we need to install complete

Hi Once I installed astrisk , how do we connect with SIP interface ? Can somebody guide me how to integrate SIP interface with asterisk ? I want to use Astrisk just for IVR purpose. Thank you Luke -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130511/a6d32bfe/attachment.htm>

Hi all, I am a fairly new * user, so please bear with me. I have been scoring the net for this info, but have had little success. I am looking for any tips and info on how to build a "information delivery" ivr app. By that I mean it will simply play prerecorded sound files and respond to key presses. Basic idea is this, 1. answer call 2. go through various standard menu's etc

Hi? As I seen on the Asterisk web site , there is packages called ,? AsteriskLatest Version - 11.4.0 asterisk-11-current.tar.gz?and? asterisk-1.8-current.tar.gz May I now which one is the most suitable for a production environment ? Thanks in advance Luke -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I encountered a DTMF problem. We have an IVR built on Asterisk 1.0.7 with RedHat 9. When the caller calls into our IVR, and IVR plays the first prompt and asks caller to dial four-digit extension. Caller has to dial slowly, otherwise, Asterisk cannot recognize the extension number. I look at the trace on Asterisk CLI and there are missing digit in the middle of string. ex, caller

REPOSTED with MORE Info and Modified Subject Line: -------------------------------------------------------- I am using one of the Minute Provider to dial out USA numbers. Now in one of my process, we need to Dial IVR and the enter DTMF digit and then it connects to the automated IVR. When I dial out the IVR directly using Xlite and VOIP Mins provider , it works perfectly. but when In try from

I' got a problem, using asterisk-rc2 :IVR functions (Background...Playback...etc) doesn't works : Executing Background("OH323/RXXXXX", "vm-extension") in new stack channel.c:1650 ast_set_write_fornat: Unable to find path from GSM to G729A---Asterisk box supplied only with network adapter.---Asterisk box registered in Mera (soft-switch with H323 protocol) and doing

Hi list, I hope this isn't in error but if it is I apologize. I have a small project request on hand where the clients want their customers to be able to dial in to conduct business over the phone in a completely automated manner. From my limited understanding this looks a lot like a call center where one has to build some sort of proxy that understands their business logic and that can

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:

How do I get asterisk to populate the "Calling Party Number" field in an H.323 call? I have asterisk configured to accept a SIP call and connect it to an H.323 IVR system. The call goes through, but the caller id is put in the "Display" field rather than the "Calling Party Number" field. -----Original Message----- From: Skuse, Phil [mailto:Phil.Skuse@vicorp.com]

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 =>

Hi, I am working on a H.323 project which involves GnuGk and Asterisk My current goal is to provide IVR functionality for the H.323 users which register through GnuGk(eg. call credit information) I have successfully built a H.323 platform using GnuGk - it uses SQL accounting and authorisation. Now I am trying to integrate it with Asterisk in order to provide IVR functionality as I already

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

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

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 ?