similar to: DBput from the web?

I have put together this call forwarding and dnd config: I'm sure it can be dome with macro's but I couldn't figure that out... anyone care to input. 74 Turns DND on my phone will not ring, drops caller to voicemail... 73 Turns DND off 72+ext forward your extension to another extension and voicemail is left at the forwarded extension. 71 turns off call forwarding. ; dnd Could

Dear All, I was trying to enable call forwarding, following the steps of the link on voip.org regarding this issue it doesn't work and the phone I am trying to implement on is still ringing. below is my conf in extensions.conf and the CLI output during the process. My configuration is : exten => _*5X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:2}) exten => _*5X.,2,Hangup exten =>

Hello I upgraded to CVS head yesterday (due to the lack of zaptel drivers working with 2.6.10) And noticed that now DBGet and DBPut have been deprecated in favour of the new Set/DB one. In the UPGRADING.txt in Asterisk it says: * The applications DBGet and DBPut have been deprecated in favor of functions. Here is a table of their replacements: DBGet(foo=family/key)

Yo, Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate call divert-feature. This one validates if the extension a call-forward is to be set to is actually valid for the current context and additionally saves this context into the DB and always uses it to originate the divert from, as you can't expect the

hi, This question is about DBPut and DBGet, Can i put about 1000 keys in a single family, (only once for the lifetime) for ex. exten => _X.,5,DBput(family/key1=${val}) ... exten => _X.,5,DBput(family/key1000=${val}) like above and if i later retrieve it, randomely, with inbound calls, will it affect performance? Surajee -------------- next part -------------- An HTML

Where can I learn the syntax for DBput and DBget? is it working with MySQL? do I need to set up tables? URiel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030508/a2598dc8/attachment.htm

hi just donwloaded the 1.6.1 branch and made configure & install. so far so good. after staerting asterisk with: asterisk -vvvvcr Could not load features.conf == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls == Manager registered action Park == Manager registered action Bridge == Manager registered

Auto Callback on Busy Register on Busy I have implemented it as 1- I store Caller and Called party numbers in database when Called part is busy 2- I retrieve it from database and Caller is called by called party when Called party hangs up It is working fine with all kind of SIP phones I have with me basic configuration for extensions.conf is given and can be accommodated according to

Ok, first off, Asterisk is the coolest piece of software I have EVER had the pleasure of using in my 8 years of running linux !! and I know I haven't even scratched the surface feature wise. Before I get too excited, I wanted to get all you experts to look at the how I implemented my after hours test. The goal is to prevent the phone from ringing afer certain hours, just go to VM.

Yo all, As there has been some intrest, here's my updated version: I post it to "-dev" as well as "-users", as it may be of intrest to both. Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate set of features. Currently, my implementation supports call- forward unconditional, on no answer

Hello, I want to enable call forwarding for asterisk 1.6.0.6 I couldnt seen any config or option on gui or extensions.conf about it. I found some dialing plans to enable it on web as follows: [apps] ; Unconditional Call Forward exten => _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4}) exten => _*21*X.,2,Hangup exten => #21#,1,DBdel(CFIM/${CALLERIDNUM}) exten => #21#,2,Hangup ;

Hei! I have a little problem with the subject. I use Asterisk CVS-HEAD-09/06/04-12:42:56 as a production *, but I do tests with a newer version Asterisk CVS-HEAD-11/18/04-10:01:32. Ok the problem is: in extension.conf I use macro for redirection, found on wiki pages: [macro-stdexten] ; ; Standard extension macro (with call forwarding): ; ${ARG1} - Extension(we could have used ${MACRO_EXTEN}

This is a follow/find me script that I can't quite get to work, asterisk wont save forward/${calleridnum} to AstDB... any comments or thoughts on how to make this work or change it to work differently are appreciated. The voice prompts to go with all playback/background extensions are commented appropriately. I hope this code is of use to some of you and any help with a perfected

I set up a call forwarding script in extensions.conf which uses the AstDB but I'm wondering if I reboot the server, will the entry in AstDB still reside? What the script does is when a call comes in, it check to see if there is a null value or a call forward number. If null, it will call the local office connections. If there is a number, it calls that. Now I just need to know if I reboot

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hi, Thinking about an IVR application and trying to get a handle on the best way to structure it so that the maximum number of concurrent calls can be achieved.. If the voice prompts were stored in a GSM format and were being played out through an IAX trunk that uses GSM compression would asterisk do a decompress/compress on the audio or would it simply pass through the GSM encoding?

Hi there How do I change the dialplan runtime, if I for example wants all calls on the main number to be answered by a voicemail (when it is out-of-office hours). I want to be able to change the configuration by pressing a DTMF combination e.g. *82. Can't figure out whether it is necessary to change contexts or how to do it. I have read a lot of examples and config documentation, but I

Has anyone implemented "hot seating" in any neat way? This where people can log in to any phone in the company and have their calls/voicemail come to that particular handset.....

Greetings fellow list members, I am trying to add some tricky functionality to Asterisk dialplan and I was curious if anyone else has come up with a solution to something like this. Basically I have phone representatives that log into one of several queues (not using chan Agent, we log in by the extension), and frequently these agents have to make attended transfer calls to outside numbers.

Andi - I have experienced the same issue you mention and gotten no reply as to a way to fix it. I finally implemented "blacklist" into my Asterisk and added "Anonymous", "anonymous", "unknown", "Unknown", etc., into my blacklist file. When those come in with an IP address instead of a phone number but have no real name, they get the