similar to: Configurable auto forward in Asterisk

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

Is it possible using asterisk to allow someone to dial in and remotely change where their call is forwarded to? For example, I'm working from home so I want my calls to go to 555 1234, now I need to go out for a bit so I'd like to phone the office and using DTMF tell the asterisk PBX to now forward my calls to my cell phone 555 3456 Has anyone implimented anything like this? R.

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

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)

I'm having trouble finding documentation for the auto attendant does anyone have an idea where there might be some???

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

Hello, I've managed to build in the "last number repeat" outlined at http://www.voip-info.org/wiki-Asterisk+last+number+repeat to call back the last person _I_ called from a particular phone, and now I'd like to try to do something similar for the common *69 -- call back the last number that called me. I assume I'll do part of this in my standard extension macro --

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

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

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 =>

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.

I was wondering if the following was possible: 2 separate incoming contexts. The first will be used when there is a secretary present. The second will be used when there is no secretary. I know that this can be done using includes and specifying the time in which each separate context would be included. However, I would like to be able to switch them from the reception telephone. For

Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM

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, 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 ;

Hi ! I'd like to connect phpgroupware to asterisk: when a user click on a phone number, his phone rings and he gets connected to the number he just clicked. I've tried by putting various files in /var/spool/asterisk/outgoing, without results (we are using SIP phones + CAPI channels). Is there a way to do that ? (If it's impossible (something impossible in *, LOL ?!?) I will create

My setup : Trixbox 2.6.1 & TE410P running well .: 1. I need to store the CallerId of the PSTN caller with his language preference so that next time he is played the prompt in his language that he chose the first time.What would be better - storing his number in the Asterisk DB and using Dbput and DBget ? or storing it in MySQL from the dial plan and quering it everytime to see the callers

Hi All, I am using a Grandstream BT100 and I have been trying to get the PBX features to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of

I'm trying to figure out a way to make my own agent login, because I don't like how the default works. I have the login and logout working fine using the dynamic add and remove commands, but I need to be able to create a list of users and passwords. I thought of a way to do it using a list of passwords, but the agent would only ever be prompted for their password. I won't want that.

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?