similar to: Can I disable musiconhold for agents

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 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

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

Is there a way to record your own voice messages ("Welcome to my PBX, Press 1 for ...") using asterisk and an analog phone, or do they need to be recorded using traditional voice recording software? Also, I am confused as to why my replies to the message board are never indented. Call this the ultimate newbie question, but how should the reply be worded so that I don't screw up

Here is the situation that I would like to create: Call comes in Receptionist sees that the caller ID is Jenny <8675309> Receptionist picks up phone and transfers call to Batman Batman looks at his phone and sees that the caller ID is Jenny <8675309> I can't seem to figure out how to forward the caller ID. Is this possible with Asterisk?

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

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?

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

This morning, I received a very strange error message on the Asterisk console. The error occurs when I try to access iconnect WARNING[196621]: File chan_sip.c, Line 386 (__sip_xmit): sip_xmit of 0x80d0854 (len 649) to 213.137.73.178 returned -1: Bad file descriptor I also get this error when I try to reload: WARNING[16384]: File chan_sip.c, Line 5355 (reload_config): Unable to get IP address for

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?

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?

Ok. I am confused. I now have conflicting answers to my question: Do you need to use a special phone to use SIP? My setup is X100P and TDM10B. I would like to connect to iConnectHere, which uses SIP. Has anybody done this before (using similar equipment to what I have listed above)? And if it is not possible, could somebody please explain why. I don't understand why this wouldn't

I am always getting multiple PCI Master Abort messages when I try to plug in a second TDM400P. I have asked this question before, but I nothing really solved my problem and I just put it on the back burner for a while. I am at a point where this is a crucial issue. I have read that the Zaptel devices share an IRQ, is this causing the problem? Is there a way that I can manually change the IRQs of

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