search for: telantek

I am working with the telantek.adsi file, and I was wondering how I would create a softkey for Transfer. I tried making a key definition and using SENDDTMF "#", but that didn't work. Is there another way I could do this? Also, does anybody have any ADSI scripts for use with Asterisk that they would like to share...

...Press Flash/Callwait on phoneB Press 700 to park the call A voice says that the call is parked at 701 When I try to dial 701, I don't get connected to the parked call Below is the asterisk output when I tried to park the call. Zap/5-1 answered IAX[telantek@x.x.x.x:5036]/2 -- Starting simple switch on 'Zap/5-2' -- Started three way call on channel 5 == Parked IAX[telantek@x.x.x.x:5036]/2 on 701 -- Playing 'digits/7' -- Hungup 'Zap/5-1' == Spawn extension (mycontext, 1003, 1) exit...

...cess 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 BusinessOne.telantek.com, SIP disabled I have not changed anything in my sip.conf file recently. Here is what I have: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls allow=gsm allow=ulaw allow=...

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

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

I want to edit my dialplan to change messages depending on whether or not it is a holiday. I have found only one example of this, but the syntax is not explained. The example is: include=>holiday||||1|jan| I know that this would use the holiday context on the first of January, but why are there 4 pipes before 1 What is the exact syntax of this? Let's say that I want to use the holiday

Newbie question alert! I was just wondering how a receptionist phone would work with Asterisk. (I've never had a real job, so I've never really looked at different phone systems). I have looked around on the internet and seen that you can purchase 24 line phones; how does that get connected? What kind of wiring goes to the reception phone? How would I go about adding one of these to an

Is it possible to have both a busy and an away message when the call waiting feature is enabled? extensions.conf ... exten=>403,1,Dial,Zap/3|10 exten=>403,2,Voicemail2,u403 exten=>403,103,Voicemail2,b403 ... Because I have enabled call waiting, I can't see how it will be possible to get the busy message to play (because there will always be a dial tone). Am I right, or do I have

I have found out why I am having problems with the directory application. The problem is that the Directory application is not searching through contexts properly. Example: extensions.conf [default] exten=>*,1,Directory(company) exten=>556,1,Directory(default) exten=>555,1,Voicemail2,u555 ... [company] ;Batman exten=>401,1,Dial,Zap/5|15 exten=>401,2,Voicemail2,u401 ***

I want to be able to check my Bell voicemail (*98) using a phone attached to my asterisk box. In extensions.conf I have defined exten=>*98,1,Dial,Zap/g1/BYEXTENSION However, when I dial *98, I just get a fast busy signal. Is the * digit reserved for internal purposes? Any help is appreciated.

This is kind of a repost of one part of a previous question I have had. Peer Username Call ID Seq (Tx/Rx) Lag Jitter Format 213.137.73.178 xxxxxxxxxx 3705df0a5f7 00103/00000 00000ms 0000ms 4 1 active SIP channel(s) I see that there is 0ms Jitter set. How can I set a Jitter buffer for use with sip channels? I can't seem to find any documentation about this.

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

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?

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

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?

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?

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

One problem I noticed when I was testing was that if I use the flash button twice in a conversation, asterisk will crash. Does anybody have no idea why? MusicOnHold Problem: I'm really not understanding that feature. I know it works, because I can set up an Extension that plays the musiconhold music. ------------------------------------------------------------------------ ----- Somebody

I am still getting a segmentation fault when I try to reload asterisk Here is the output: Asterisk Ready. *CLI> reload == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/enum.conf': Found == Parsing '/etc/asterisk/rtp.conf': Not found (No such file or directory) == RTP Allocating from port range 5000 -> 31000 -- Reloading module

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