similar to: More Level QueueSystem

Hello Experts, I have a severe Problem with my current samba installations. Here in my Home-Office networt I have a dual-CPU machine with Debian Sarge, hosting some samba 3.0 shares and as well being a host for 4 vmware-sessions (Vmware Workstation v4.52, with 3xServer2003 and 1xXPpro). As well I have a separate raid5-server with samba 3.0 in the exakt same configuration (of course all parameters

I'm Japanese. Sorry,English is not so understood,Please let me question by items. In Asterisk-1.2.1 and 1.2.2,I cannnot understand the operation of Authenticate application's 'j' option. exten => 123,1,Answer() exten => 123,2,Authenticate(789,j) exten => 123,3,Playback(pin-number-accepted) exten => 123,4,SayDigits(111) exten => 123,103,SayDigits(999) In this

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am having problems with the PlayTones application and VoIP softphones. I have the following in my extensions.conf: exten => 123,1,Answer exten => 123,2,PlayTones(Busy) exten => 123,3,Hangup But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. I get the following on the console: --

Hello Experts, I have a severe Problem with my current samba installations. Here in my Home-Office networt I have a dual-CPU machine with Debian Sarge, hosting some samba 3.0 shares and as well being a host for 4 vmware-sessions (Vmware Workstation v4.52, with 3xServer2003 and 1xXPpro). As well I have a separate raid5-server with samba 3.0 in the exakt same configuration (of course all

Hello Experts, I have a severe Problem with my current samba installations. Here in my Home-Office networt I have a dual-CPU machine with Debian Sarge, hosting some samba 3.0 shares and as well being a host for 4 vmware-sessions (Vmware Workstation v4.52, with 3xServer2003 and 1xXPpro). As well I have a separate raid5-server with samba 3.0 in the exakt same configuration (of course all parameters

Hi, I would like to develop a dialplan that allows the callee to reject the call like this:- 1) Call comes in and receives a greeting and get put into a queue. 2) A second call is placed to the member of staff (SIP phone or mobile phone) 3) The member of staff answers the call and is presented with a few options. 4) If the member of staff presses 1, the incoming call is connected to the member

Greetings, I have a dialplan that calls the dictate application, but I want to do some post-processing on the RAW file created. The post processing is working fine as long as the dictation application exits gracefully, but fails when the user simply hangs up. How can I make sure the system() command is run regardless? Example: [test-dictation] exten => 123,1,Dictate(/tmp/dictate) exten

Hi everyone I am having problems with my call queue We currently run a customer care call center which has attendants login during the daytime. Customers who call the 'customer care line (a specific number) always get routed to the cutomer care queue (called 124). After hours, staffs of the Network operating center provide customer care services for customers who call in after the last

Hi all, There is a problem with my dialplan. here is the dialplan: exten=> 123,1,Dial(SIP/U1,,Ttg) exten=> 123,2,Hangup exten=> h,1,AGI(onhangup.pl) The problem is whenever U1 is called or calls someone, if U1 hangsup the call then the h extension is NOT executed. but if the other person hangsup the call, then the h extension is executed (assuming that the other person is calling

Hello, I'm just starting out with asterisk and I'm playing around with the system. Currently I have a Digium TE210P connected to a PRI on the Asterisk server. I have a SIP soft phone on my laptop for testing that is working fine. When I try to place a call from my soft phone I get this from Asterisk: May 1 09:11:41 NOTICE[20098]: app_dial.c:1029 dial_exec_full: Unable to create

Hi, I am setting up a hylafax server. From what I've read so far, hylafax supports CID numbers and names but currently does not support DID. I assume I can do something like this... [40faxDIDs] exten => _87[5-8]X,1,SetVar(CALLERIDNAME=${EXTEN}) exten => _87[5-8]X,2,Dial(Zap/g${hylafaxMODEMGROUP}) ...and use the CIDName variable in hylafax to route the faxes to the appropriate

Hello all, I have a very basic question but I haven't found any answer. I would like to configure asterisk so that it wil not indicate a call waiting to a SIP phone if it is already on conversation (off hook). But I don't want to loose call transfer, call hold and so on. Is there any possibility to do that? Regards, Daniel ANDRE -- Daniel ANDRE (mailto:daniel.andre@iris-tech.fr)

Hi, I am using asterisk1.2 In this, my dialplan is going large , so i need to configure this small pieces for this, i did in my extensions.conf when I dial the 123 its not going , means that file is not reading. is there any parameters to add any where ? please tell me this #include is not working ... extensions.conf [general] [global] trunk=zap/g0 #include exten-internal.conf [default] exten

hi, i have a new install, 1.6 2, 2 extension, but the call doesnt get thorugh: here is my sip debug outout: thx for ur help!! <asterisk-users at lists.digium.com> --- (13 headers 16 lines) --- Sending to AA.BBB.CCC.DD : 28127 (NAT) Using INVITE request as basis request - Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4NjAyOTYzMjk. Found user '701' for '701' Found RTP audio format 107 Found

It can be done. 1. Setup a new Vm profile on CCM with a mask of XXXX 2. Setup a CTI route point: a. Set the directory number to a pattern. I use *27XX but any pattern that you can send from * is good, ie. 88XXX b. Set the VM profile to the newly created profile c. Set the line to forward all calls to VM 3. Change the dialplan in * to append the extension called to the

Hi, I've got a CCM ( Cisco Call Manager ), with a Cisco Unity VM server and about 45 SCCP phones on the ccm, and 200 users on unity. we want to migrate all users to IP Phones to ditch our ancient phone system. I would love to get Linksys-Sipura SPA-941s for the 150 users not on IP phones yet and run sip to an asterisk server, but have their voicemail on Unity. these phones are $150 each,

Hi all, I just started programming using AGI and I have a simple question about STDERR. If I understood it right, all the messages sent to "STDERR" should be shown in the Asterisk console, but using the following python code I just can't see anything. #!/usr/bin/python # # File: /var/lig/asterisk/agi-bin/agi-test.py # # Description: An AGI Script # import sys env =

I installed Asterisk 1.0.9 in a Freebsd 5.4 ( with no zaptel card); I have 2 zoom x5v and works great ( in extensions 123 and 321 ) but I was trying to test cmd Playback, MusicOnHold, MP3Player but when I call to extension 100 I don't hear the sound ( mp3 or gsm that I put) , I only hear noise If I leave a message in a mailbox the same, all the record is noise --------- extensionns.conf

Hi all. I posted this earlier but never got any advice that helped. If anyone knows how to get this going, I'd appreciate some advice. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten =>

Hi there, i have a Problem with the Pickup command. Versions: asterisk 1.4.1 on gentoo my extensions.conf [only the interesting part]: [incoming_1] exten => 123,1,Ringing exten => 123,2,Dial(SIP/xxxx,20,r) exten => 123,3,wait(90) exten => 123,4,hangup [incoming_2] exten => 456,1,pickup(123@incoming_1) both are sip-accounts and have pickupgroup=1 in the sip.conf so my idea is,