similar to: Newbie :Call Forwarding problem

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?

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

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 ;

Does anybody has a php code for using DBput (DBget, DBdel) from a web interface, which database is used for astrisk? bye Ronald

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 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.

I found an unanswered mail in the archives that implied that perhaps there is no direct way to delete a DB entry with the new "Set" syntax. I have been playing, and so far if there is a way to do it I haven't been able to suss it out. Set(DB(family/key)=) sets the value for the key to null, but that doesn't appear to be equivalent to removing the key entirely. Or maybe

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 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

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

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

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}

Hello, I'm trying to do call forwarding based on this: http://www.voip-info.org/wiki/view/Asterisk+call+forwarding In the extensions.conf file I have several context defined (local, longdistance, mobile, international and so on). Each user can be associated with different context (so can make only i.e. local calls). How to set calls forwarding only to numbers that are available in

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.....

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

hello all, i need some help and after trying the wiki i'm even more confused than i was. i'm trying to set up call forwarding and running into problems... i want the most basic call forwarding imaginable. 1. caller dials extension (say, 154) 2. dialplan is updated to forward caller's extension (based on CALLERIDNUM) to voicemail, instead of ringing his endpoint. 3. caller is

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)

Hi, I'm trying to setup a small BRI ISDN <-> voip gateway. The ISDN card is based on Cologne chipset, so I try set it up with zaphfc. The versions i'm running: kernel-2.4.27 Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e zaptel modules 1.0.7 zaphfc is from bristuff-0.2.0-RC8e When I'm doing the insmod on zaptel, zaphfc, zaprtc: Zapata Telephony Interface Registered on major 196 PCI: