similar to: Can I have same numbers in different contexts ?

Hi Folks, have a question, on bandwidth. I want to run an asterisk server SIP to H323, g729. Calls arrive on sip/iax go to IVR get authenticated and egress through h323. So G729 license is only used during IVR and then it is pass through. I am collocating this server. Colo offer a monthly bandwidth quota. Lets say I want to do 100K minutes per month of VoIP calling at the beginning. What would

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

Folks, I want to build a custom IVR for my setup. I've got it working (well, the bells and whistles are not there yet, but the basic stuff works) using AGI, but I'm worried about how well this will scale. I've seen references to FastAGI, and presumably this will be more efficient. Question, though: how well do either of these (AGI or FastAGI) scale if my system is handling a large

Hi All, We are just starting to deploy a new PRI IVR system, and the incoming calls sometimes get random short 'bips' while navigating our IVR menu. Any hint on what this can be? Best regards, PauloHM

Hello all ! I have a problem with my IVR with terminate connection from PSTN phone Here is my configuration extension.conf [ivri] ;exten => s,1,Wait(1) exten => s,1,Answer ;exten => s,2,DigitTimeout(5) ;exten => s,3,ResponseTimeout(10) exten => ivr,1,Background(demo-congrats) exten => 1,1,MP3player,/mnt/linux/mp3/song/04.mp3 exten =>

Thanks to this group, I am making great progress in getting up and running with Asterisk! I actually connected with a SIP client last night and made a phone call (still getting some error reports from Asterisk -vvv during the process and intermittently regarding SIP registration errors). Up to this point, I have been using the .conf files that were created during the basic compile and install

I want to be able to demo * on a notebook at a client's site. This means no FXO gateways; just 2 sip phones (like SNOM) and maybe a softphone (GnoPhone?). I already have RH9 running on my notebook. I would like to have one SIP phone dial and go through IVR before making a choice and ringing the other phone extensions. Of course the notebook would have to be running Asterisk. How can i setup

Hello, It is my understanding that on the softphone side, asterisk is only responsible for establishing the session between two phones. If this is the case, does it matter what type of audio codecs the two phones are using? And if it does matter, are there any codecs that cause problems with asterisk bridging two SIP connections? Thanks for your helpful input, Daniel

I'm in the process of reworking my dialplan to include an ivr and other items. I've seen several examples over the last several months that mention the "s", "h", "t" (and probably others) extensions, but I don't fully understand what they are used for. Can someone either give a short definition of each or point me towards some doc?

exten => auth_dial,1,DigitTimeout,5 exten => auth_dial,2,ResponseTimeout,15 exten => auth_dial,3,Read(dialed,IVR/en_enter_destination,0) exten => auth_dial,4,agi(call_start.php|${dialed}) exten => auth_dial,5,dial(SIP/${dialed}@146.82.15.241) I'm trying to get What they dialed put into the PHP script. How do I get the contents of this variable in the php script?

Hello all: Thank you for taking the time to read this post. Background: I am a new user to IVR systems and asterisk. I have been tasked with helping to set up a system that will only handle IVR (eg no PBX functions) incomming calls for 45 or so people that will call in 3 or 4 time each day during (approx) normal business hours. We have started to look at the Ivrs perl module from

Hello list, I want to listen to voicemails on my * box from a phone that is not local to my pbx. I.e., from my cellphone or my PSTN work line etc. I'm aware that I can forward VM to email or use a web interface but that is not always practical. Other than doing an IVR type arrangement or a phone number dedicated to VM access is there a way to do this? On my old POTS line I used to be

(Embedded image moved to file: pic18467.gif) I am a collaboration consultant doing some research for a major client who has a problem. They have an existing IBM Direct Talk 2 IVR which they were going to upgrade to the current Websphere product, but choked on the price. My task is to find out the following: 1. The spec calls for a 24 analog line system with a fairly sophisticated response

Howdy, I need some pointers (ideas or help) to build a solution to retrieve recordings out of an IVR. The program needs to do some predictive dialing functions I only need it to: 1) Be able on it's own to make a call (to the same number inside this script). 2) Detect that the call has been answered and that it has finished. I need some sort of AGI but it needs to run on it's own, i.e.

Has anyone else had trouble with the AGI command GET DATA on the latest stable cvs? I can't get it to work with asterisk-perl, or by using print statements and reading stdin. I get "200 result= (timeout)". (this is from the print statements, and asterisk-perl reports nothing). But asterisk is getting DTMF because my menu in extensions.conf works. I will go through the code

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

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

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

Hi, I'm looking at setting up a small production system - predominantly for voice mail and IVR (with a few extensions and hold music MP3's). I've found a couple of IBM X330 servers, with dual 1.13Ghz P3 processors. My question is; is a dual 1.13Ghz P3 server sufficient to run for real-life demands? I come from a Unix/Mac background, so I'm not swayed by the '3Ghz'