search for: hecken

...hy? Through the include statement, asterisk has to look first in capi-in-sub, there it should find this extension: exten => 95562315,1,DoSomethingAnybodyWouldExpect() ... and follow the dialplan under capi-in-sub since a valid extension was found. What's wrong, any ideas? Regards, Guido Hecken gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 K?nigswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:guido.hecken at gwsnettech.de

...nough to let the user "find" their keys on the phone, the double ammount of time ( 2 x 5 secs) you have to wait before a call is parked and the parkposition is announced, is really too long. Did I miss something in the documentation? We are using SVN-branch-1.4-r96449. Regards, Guido Hecken gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 K?nigswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:guido.hecken at gwsnettech.de

...kup=no in sip.conf but it made no difference. > > Does anyone know how I to make Asterisk continue working for local LAN > users/gateways when a DNS server is not reachable? Try to use bind on the * Machine and configure it as a caching only nameserver. Hope, this helps Regards, Guido Hecken

Hello, I am trying to setup an Asterisk GUI with the help of astman(please visit http://astman.sourceforge.net/am-user-guide.html). I have installed astman and currently assessing my GUI using; http://ipaddress-of-asteriskbox/cgi-perl/am-main.pl I am trying to get the menu options in my GUI to work but to no avail. Currently my parameters are set to; Asterisk Install Directory:

Does anybody have any experience using a Patton SmartNode as a SIP/Telco gateway with Asterisk? They seem really inexpensive and appear to support all of the necessary features, but I don't have any experience with their products, so I don't know if they are any good. We are currently using a Cisco 2600 w/ PRI card and it works fine, but I was looking for someone else as a possible

hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar

I had the same problem with snom 190 phones. Using the transfer with # instead of "Transfer Button on the phone" worked for me. In my configuration "REFER" was not send, so the transfer with the button on the phone did not work. Guido Hecken -----Urspr?ngliche Nachricht----- Von: steve szmidt [mailto:steve@szmidt.org] Gesendet: Sonntag, 26. Dezember 2004 17:14 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] Cannot transfer after queue agent picks up call I have not been able to find anything that relates to this probl...

...663 in every line of the file, named number. But how can I achieve this in asterisk dialplan? I did some googling and reading in the wiki, but still no glue. Perhaps with the system command and some perl stuff as above? Any hints are *really* welcome. Regards, and have a nice day outhere Guido Hecken

...in features.conf: [general] parkext => 800 parkpos => 801-805 context => parkedcalls parkingtime => 60 ------------------------------------------------- We are using SVN-branch-1.4-r96449 and other, older versions of asterisk Regards Guido gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 K?nigswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web <http://www.gwsnettech.de/> http://www.gwsnettech.de <mailto:guido.hecken at gwsnettech.de> mailto:guido.hecken at gwsnettech.de -------------- next...

...allow=gsm register => 081503:xxxxxx@sipgate.de/081503 language=de tos=0x04 [6301] type=friend username=6301 secret=xxxxxx host=dynamic disallow=all allow=alaw allow=ulaw allow=gsm dtmfmode=rfc2833 mailbox=6301 context=teilnehmer callgroup=1 pickupgroup=1 group=2 callerid => "Guido Hecken [gwsNetTech]" <6301> Hope, this helps... Guido Hecken > -----Urspr?ngliche Nachricht----- > Von: Jason A. Crome [mailto:crome@devnetinc.com] > Gesendet: Mittwoch, 16. Februar 2005 17:05 > An: asterisk-users@lists.digium.com > Betreff: [Asterisk-Users] Can't connect...

Hello, I have SER and Asterisk running on the same box. I want SER to listen on port 5060 (it is) and Asterisk to listen on port 5062. I have configured my phones to register with x.x.x.x:5060 (SER) and Asterisk will purely act as a voicemail server at the moment. However I cannot get Asterisk to listen on a different port. It is my understanding that I just need to set the port in sip.conf

...d the page. See some > of the output below; Try a simple php-script in this directory. Something like this, name it test.php <?PHP print (test) ?> If it does not print "test", and you see the code instead, check your httpd.conf for a general php-parsing problem. Regards Guido Hecken

I'm having a problem getting my queue to function as it should. After 20 seconds or so, it should prompt the user with a message "thanks for holding..... press # to leave a message or stay on the line to continue holding". I set up the "context" in the queues.conf file, so if a user presses a digit, they should be able to leave. But I get a SIP BUSY message. Here are my

For anyone using CVS HEAD, if you are using queue member persistence or agent persistence, your next update will cause the persistence to break. The storage format for these elements has been changed so that it can be more easily extended in the future, but this required breaking compatibility. This should be the last time these features will be broken by an upgrade :-)

...4002-4563 acknowledged Then I can transfer the call... Weird because i?m using ackcall=NO in agents.conf ... Diego Magalh?es diego@redetaho.com.br +55 24 22335134 -----Mensagem original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Em nome de Hecken, Guido Enviada em: quarta-feira, 2 de fevereiro de 2005 15:18 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: RE: [Asterisk-Users] AgentLogin / AgentCallbackLogin transfer pro blem Which kind of transfer do you use? Try using the # transfer. Hope that helps.. Guido Hecken...

Hello all. I'm having an odd problem getting * and sipgate to work together. From Sipgate support I have gotten this repsonse to my query: ===== Your Asterisk is registering incorrectly with our servers. It registers like this: sip:s@217.XXX.XXX.XXX:5076 The "s" should be your SIP ID. Anything else is rejected. I don't know where you can find this setting, but from our

...hat drops a test call in /var/spool/asterisk/outgoing and if this call wasn't processed successfull the script stops all running asterisk and moh processes and restarts asterisk? Any help would be appreciated, since "I can't get no sleep with these timebombs out there" ;-) Guido Hecken

I just setup an Asterisk system on a small Shuttle box; I am only using SIP channels and have no FXO/FXS cards. The system works fine in that I can call my inbound number (Broadvoice) and have the system answer and I can make outgoing calls. The problem is that every time I want to change something in the sip.conf file, I have to do a 'restart now' instead of a 'reload' or

Hello everyone, I am trying to install windows xp as a guest OS. It runs through the installer just fine, but every time I reboot, it keeps booting back to the installer. Here is my config: ------------<code>--------------------------------------- kernel="/usr/lib/xen/boot/hvmloader" builder="hvm" memory=512 device_model=''/usr/lib/xen/bin/qemu-dm''

We have a large install of 360's running rev 4.1 with zero problems. I did another, smaller install couple weeks ago with 40 360's running rev 5.3. In both cases, the install was identical, same Asterisk version, same dialplan, everything the same except the differences were: 1. Different firmware rev 2. Different physical LAN Guess what? On the smaller install, lockups and reboots.