similar to: CallerID forward???

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, Ok, here is the setup. Asterisk conected to a PRI line (23 lines). 3 tool-free phone numbers are routed to this PRI line. Customer wants to have a way to have shown on the receptionist phone that the call comes from which of the 3 tool-free lines. Possibly display on the phone that the call comed from tool-free number 1, 2 ou 3 or even better a name or text id associated with this

Ok. I just got a TDM10B and it is in with my X100P. So as it says in the provided instructions, I used the command modprobe tor2 I get an error message saying that there is no such device. My zaptel.conf looks like this: fxsks=1 fxoks=2 So I load the X100P first. (modprobe wcfxo) Then I load the TDM10B (modprobe tor2) Then I'm told that the device doesn't exist. Please help

So I have an X100P, and a TDM10B both working (at least I think they are). The drivers have been loaded and ztcfg -vv shows no errors in the configuration of two channels. When I run asterisk -vvvc and pick up my phone (plugged into TDM10B), I don't gear a dialtone. in phone.conf, I have [interfaces] mode=dialtone format=slinear ... Shouldn't that produce a dialtone when I pick up the

Guys. Anybody using sipuras 2002 knows if there is a way to make the phones connected to it to autodial an extension when the phone is picked up? For example, if the phone is on a police booth (building entrance) and you want the guys to just pick up the phone and make the phone auto dial the receptionist extension without the guys having to dial anything (ala batphone). Is this possible with

Here's the problem: I start asterisk, and it takes up around 3-4% of my CPU resources. However, this number continues to climb over the hours until it is close to 100%. Usually it takes around a day to climb up to approximately 95 or 96% Has anybody experienced the following problem before?

I was playing with the agent application to see if I could get it to work. Everything works fine, except that Asterisk plays musiconhold while an agent is logged in and is not taking a call. Is there a way to disable the music in this situation? Imagine working tech support where you had to listen to hold music when you weren't taking a call. Now think of your company's choice of hold

My setup: X100P and Quicknet PhoneJack. I can't seem to properly set up a Zap channel for my X100P. Here are some of my configurations: [zaptel.conf] fxsks=1 #X100P fxoks=2 #Quicknet PhoneJack defaultzone=us loadzone=us [zapata.conf] [channels] context=local signalling=fxs_ks channel->1 ;X100P [extensions.conf] ... [local] exten=>_NXXNXXXXXX,1,Dial,Zap/1 ;I'm pretty sure the

I need help creating a channel for my phone device (Quicknet PhoneJack). I have installed and loaded the driver and phone devices listen in /dev (phone0 - phone15). [phone.conf] mode=dialtone format=slinear device => /dev/phone0 fxoks=2 ;Quicknet PhoneJack [extensions.conf] ... exten=>_NXXNXXXXXX,1,Dial,Phone/phone0 ... When I try to make a call, I get the following output: Executing

If extension 101 calls 102 and user 102 hits # and then 103, the caller ID of 103's phone says 102. I've been looking for a way to have 103's Caller ID show the person that is being transferred not the person transferring. So if my receptionist answers the phone and transfers it to one of my techs, I want my techs phone to display the caller ID of the person who called the

I just wanted to try out Festival, but I can't get it patched. I'm thinking that there is something missing from the steps listed at http://www.marko.net/asterisk/archives/0209/0389.html. >>tar xvzf festival-1.4.2-release.tar.gz >>patch -p0 </usr/src/asterisk-ng/festival-1.4.2-diff >> (or wherever the patch is located) When I run the patch command, I get the

Does anybody know a place where I can download a working PWLIB and H323? The files at www3.openh323.org are not in gzip format (according to the error message I receive) gunzip: pwlib_1.4.11.tar.gz: not in gzip format I have also tried using the tar -xzvf command, and that gives the following error: gzip: stdin: not in gzip format tar: Child returned status 1 tar: Error exit delayed from

I used to have email notification working with my voicemail services but it stopped working when I installed the new version of asterisk. I have not changed my voicemail.conf file, so I'm out of ideas. Does asterisk use Sendmail to send messages, or does it have its own method for sending email?

Hello, using Asterisk 1.8.12.2 case : I call with my cellphone to our public telephone number Our receptionist answers the incoming call and does an attended transfer to my colleague ( A ) Colleague answers and the receptionist tells him that I am on the other side. Receptionist transfers the call and I am connected to my colleague ( B ) My question is about the CallerID that the

Ok, this falls under the newbie category: Has anybody created any documentation on using musiconhold or call parking? I have found sample config files for musiconhold, but I'm not sure how they work. [musiconhold.conf] [classes] loud=>mp3:/var/lib/asteriks/mohmp3 How do I then set up this feature in extensions.conf? I can't seem to find an example of what I'm looking for (or I

I'm having a problem defining my channels Here is my zapata.conf ----------------------------------------------------------------------------------------------------- ; Zapata telephony interface ; ; Configuration file [channels] ; rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes

I'm wondering if I can do the following: Caller activates the Directory application Caller enters the first 3 digits of a person's last name ===== Normally here, Asterisk will say the extension number of a person found. Is there a way to get Asterisk to say the name as well? (perhaps using the same sound file that is used for their name in the voicemail application) Can this be

Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that

I am trying to run the make command to install Asterisk, but I get the following error: make ... ... checking for tgetent in -ltermcap ... no checking for tgetent in -ltinfo ... no checking for tgetent in -lcurses ... no checking for tgetent in -lncurses ... no configure: error: termcap support not found I am running Mandrake 9.1 on a Pentium II 200MHz. Could this be a hardware issue? I

Besides email notification, is there another way to get asterisk notify the user that they have a message? Example: Some analog phones have a blinking light that lets the user know that they have a voicemail message. Is asterisk capable of doing this?