similar to: IVR sample config?

I'm testing an ivr implementation (first time) using: exten => 620,1,Wait,1 exten => 620,2,Answer exten => 620,3,DigitTimeout,5 exten => 620,4,ResponseTimeout,10 exten => 620,5,Background(npi-greeting) ; "Thanks for calling press 1 for" exten => 1,1,Goto(npi-directory,s,1) For initial testing, I've arbitrarily mapped this onto ext 620 (will change that later

I'm trying to setup a VERY simple mainmenu ivr but can't seem to get the tones to be recognized during the background( ) the playback and background files play, but asterisk doesn't do anything when I start pushing keys - I've tried it from softphones and pstn line phones Can anyone tell me what I'm doing wrong? Required contexts Exentions.conf below [from-broadvoice]

Hi, I am an asterisk newbie and looking around for information . I wish someone could take their valuable time off to answer my query in detail. I wish to set up an IVR system that can allow user authentication and therefter accept 2-3 inputs from users ..generate a key and transmit the same in voice back to the user . The system will intially have small load but if the whole package in future

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/104a07f3/attachment.htm -------------- next part -------------- Hello In my IVR menu whenever user select the option number 1 then it should jump to echo context, I think call did jump to "echo" context but I always get the following warning and I hear couple of beeps and then

Hi I work at a small company that has some IVR solutions that use Dialogic hardware for everything. Everything is written in C++ using MS VC++ using the Dialogic API and runs only on Windows. Being the rebel that I am, I would like free myself from Dialogic. To do this without porting all our existing code to run on Linux I was thinking of controlling the Asterisk from a Windows machine running

Are IVR and "Auto Attendant" interchangeable terms? They both do the "Press 1 for" thing. Sales is asking me how to word it and I've always used both terms interchangeably.

Just turned up a new PRI with DID's in the US. I'm receiving 5 digits of the DID numbers as I requested. Assuming I have 100 DID numbers but only define 50 of those in extensions.conf, is there an easy way to send the incoming calls for the 20 undefined numbers to a common resource (ivr, operator, or canned message) without having to define each one?

Hi, Yes InternalExtension is the context and 2093 the extension. Just to explain something odd that?s happening (and I?m very stumped with this) .I think my contexts are definately the reason that I can?t interrupt the menu for incoming pstn calls to choose a submenu: My users register with my sip proxy (SER). Therefore when I create an entry for them in sip.conf I set only one context. Also to

I've been working on an asterisk box at work for a few weeks now, things were finally starting to sail smoothly until I hit this head scratcher this morning. It's a rather intricate problem, so bear with me. Heres the scenario. What works: If I call from my sip phone -> sip phone everythings ok If I call from sip phone -> external pots number ok as well If I map one of our

Hi all, I've got some questions to post in regard to running asterisk in a production-grade environment, specifically targeting high-density IVR applications. No VoIP involved, just straight PSTN -> * and perhaps the occasional outdials or agent-based predictive dialing. 1) Which user would you run * under? 2) What other security-related issues do you have to resolve? 3) How do you handle

Just throwing this out here, hopefully someone can tell me why. *CLI> show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is

Hey all, I am dialing a DID through VoicePulse Connect. The number is answered by a main menu type of IVR. The configuration is as specified in both the wiki and VoicePulses documentation. The call comes through without a problem, but when the caller enter any keys they are either not recieved by * or they are ignored. With SIP I would typically put a dtmfmode= line under the peer and

Hi- I'm almost embarrassed to ask the following simple question, following John's excellent and rigorous bandwidth analysis (see earlier thread): I have a straightforward Asterisk application, IVR-only (no connections between channels). It will simply decode DTMF's and speak prompts (probably A-Law encoded) on a number of E1 circuits simultaneously. Realistically, how many

I have a number of Zap/ extensions defined in a queue with ringall strategy. When this queue is called sometimes Asterisk seems to think that one of these channels is busy, while it is NOT. The following is shown on the console: --Called 44 -- Called 36 -- Called 41 -- Called 35 -- Called 38 -- Zap/44-1 is ringing -- Zap/36-1 is ringing -- Zap/41-1 is ringing

Hello, I have had several instances over the last month of Asterisk freezing, sometimes after 12 hours, sometimes after 8 days. The common elements are that: - all Zap channels lock[hangups don't register and no new calls in or out] - no new in/outbound calls can be made on Zap or SIP channels - people who are still connected to calls can continue to talk - in the CLI interface, you can

Hello: I have found the following problems with outgoing calls with asterisk, compiled with an updated CVS on 22 Oct. 1.- Problem with retries: Whenever I set the MaxRetries parameter, to something greater than 0 in a call-fille, Asterisk ignores the RetryTime parameter and retries every file in the outgoing folder when a new call-file is copied into that folder. So, if I make a call placing

Hi all- It's often mentioned here that recompiling the Linux kernel may improve the performance of asterisk. My question is: Does re-compiling the kernel have any effect if there are no changes to the configuration? What parameters will typically improve performance? Thanks for any insight! Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London,

hello, i try to build a IVR for our company my problem is that the dtmf tones are not recognized by the phones i tried several phones. BUT when i call the voicemail i can navigate with all phones through the menu. I use * 1.2 here is the context: [ivr] exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 ;SAI menu -

Thank you for reading this, sorry to waste bandwidth otherwise. I am part of a US company looking for someone to setup a demo IVR system for us. I seem unable with my current knowledge to pull this off myself. The demo is the regular enter your id and validate/repeat/continue methodoligy you put up with in everyday life. I would like to have the validation and other parts done via database

Max TE410P card on an Asterisk Hello, Does anybody know the max number of TE410P/TE405P card we can put in an asterisk box? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040504/f220d3ae/attachment.htm