similar to: using # to end a number

Hello - I'm using ATA-186 devices, with RFC2833 DTMF encoding. I am having problems with routines that input long strings of numbers, in that I am getting more than a small number of double digit entries. As an example, I have a section that asks for the user to enter a call forwarding number, and then puts that number into a database. Almost always, there are double digits when the

Im trying to create a custom application via the AGI. I want to authenticate the users that dial in with a userid and pin. However, the number of digits in the PIN and userid are variable, and therefore I need to allow the user to "press enter" by hitting the pound key. How would I accomplish this in the AGI? stream_file doesnt seem to work, since it only allows one digit to be

Hi, Greatly appreciate if some one help me with the application read(). asterisk*CLI> show application read asterisk*CLI> -= Info about application 'Read' =- [Synopsis]: Read a variable [Description]: Read(variable[|filename]): Reads a '#' terminated string of digits from the user, optionally playing a given filename first. Returns -1 on hangup or error and 0

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,

[moved from -dev, as the thread is better suited for -users] At 5:10 PM -0600 1/30/04, James Sharp wrote: > > I've pretty much got the routing covered at this point, I'm just not sure >> how to get the Asterisk system to answer and give me dialtone immediately. >> Any ideas or recommendations would be greatly appreciated. > >app_disa will give answer and give

Hi, Firstly, what I'm trying to do is: * Get asterisk to pick up a SIP call via a DID * Prompt the user * When the user puts in a number, go to IAX.conf and route it according to what I've specified there, i.e Least Cost Routing, etc. I've set-up something similar to what I've found online, but it doesn't work! Asterisk doesn't pick up the call at all..... :( The files

Hi everyone! I've set up asterisk ip-pbx to implement IVR menu and encountered such a problem: when users dial the destinaion phone number and end it up with "#" asterisk still waits until timeout in WaitExten() is reached. // Here comes the context where user is prompted for a dest. number: context ivr-dialout { s => { Background(enter-your-dest-number);

On Tuesday 24 November 2009 10:57, Anton Korobeynikov wrote: > If it is just 'normal' RISCy architecture and one should not care > about ABI weirdness (this is usually true for MCUs), then the estimate > is correct. You just provide patterns for all the operations, expand > everything unsupported. After that you need to write bunch of libcalls > and first version is ready

Dear all, I would like to run a simple regression model y~x1+x2+x3+... The problem is that I have a lot of independent variables (xi) -- around one hundred -- and that some of them are categorical with a lot of categories (like, for example, ZIP code). One straightforward way would be to (a) transform all categorical variables into 1/0 dummies and (b) enter all the variables into an lm model.

Just wanted to catch everybody up. So many people contributed to the work list that I'd probably miss several if I tried to notify just them. Many people contributed funds, too, which I greatly appreciate. We were able to get the entire list done, which added up to two full hours of work on Allison's part. And that's just the recordings -- I'm doing the editing myself. Which

Source: http://www.nanpa.com/number_resource_info/vsc_assignments.html See also: http://bugs.digium.com/bug_view_page.php?bug_id=0000071 Some (which?) of the codes below are hardcoded into Zap channels only. Is there a European equivalent for this (or ITU / IETF)? Greetings, Philipp VERTICAL SERVICE CODES (US Standard) ASSIGNMENTS *00 - Inward Voice Activated Services (English) *01 -

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

On 16/11/2018 18:46, Jean-Philippe Brucker wrote: >>> +/* >>> + * __viommu_sync_req - Complete all in-flight requests >>> + * >>> + * Wait for all added requests to complete. When this function returns, all >>> + * requests that were in-flight at the time of the call have completed. >>> + */ >>> +static int __viommu_sync_req(struct

Hi, I''ve noticed an issue (or what I think is an issue) with mongrel_cluster: in the config file, I tell it to store pid files in / tmp/ and when I start it they get saved there, but when I run restart or stop, it says it can''t find pid files. Problem being: it looks for pid files in /path/to/my/app/tmp/. Is anyone else aware of this? I have lighty forwarding to pound

Hi, I am trying to find remote call forwarding feature in asterisk. I don't know is it possible or any one had already done it. SBC (local Telco) provide such feature. I can call into my voicemail number, and set the remote-call-forward to my cell or another number. It is like person can remotely manage to set the call-forward or DND to his/her extension. Can this be doable in asterisk?

One tiny thing that would be very nice to have is a default value of n=1 in the random number generator functions, enabling, e.g., runif() instead of runif(1). This won't break anyone's existing code and ought to be relatively straightforward to do. Is there anyone in the core team who would be willing to include this change? ----- Regards, Richie. Mathematical Sciences Unit HSL --

(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

I would like to know if I can set Call Forwarding on an extension but allow direct calls from the extension it is forwarded to. Example: Extension 100 sets call forwarding (all) to extension 101. All calls to 100 are immediately forwarded to 101 as expected. However, if 101 tries to transfer a call to 100 or tries to call 100 directly, it sounds "busy" because it obviously goes into

I'm running Asterisk 1.8 from Debian. I have some analog phones connected via a TDM400P. I'm testing them with these simple extensions: exten => 600,1,Answer() same => n,Festival(This is an echo test) same => n,Festival(Hang up or press pound when you are done) same => n,Echo() same => n,Festival(Good-bye) same => n,Hangup() exten

Hello, David > That's pretty optimistic, even for someone who knows what to do. It depends on the target. If it is pretty straightforward, than you can do almost all the stuff automatically. The estimate I provided was the real time for first working version for msp430 backend. If it is just 'normal' RISCy architecture and one should not care about ABI weirdness (this is usually