similar to: SV: Help on queues

Campon, mini-queues, see asterisk tips and tricks on voipinfo... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Zach A Sent: Monday, February 06, 2006 1:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Help on queues I need practical examples showing

Thanks! Do you have any suggestions on what I might do next. I have the phones, I have asterisk, and I have everything setup. But i can't get the login to work with the Polycom function. Nothing happens...and I can't find any readmes' or manuals. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com

Thanks for the quick response! I read about logrotate at voip-info.org but I didn't quite understand it. I'm no asterisk/linux expert unfortunately. First of all. What exactly does happen when I run: /usr/sbin/asterisk -rx 'logger rotate' Does it clear the file and create a new one? Can I run this manually without any interruption in the system? And what does the script do? I

This is what i found on Cisco's site: "Symptoms: Media negotiation fails for SIP calls and the terminating gateway replies with a "488" message to an Invite message. Conditions: This symptom is observed on a Cisco platform when the terminating gateway is configured with the G279B (annex B) codec and when the Session Description Protocol (SDP) for the incoming Invite message

Im curious. Does anyone have experienced echo-problems that later where solved by buying a hardware-echo canceller such as the Wildcard TE411P? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r James Harper Skickat: den 6 februari 2006 11:46 Till: Asterisk Users Mailing List - Non-Commercial

My suggestion would be the one-line eyeBeam phone under development. Check out support.xten.com. //Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r afoc@interconnessioni.it Skickat: den 18 oktober 2005 14:48 Till: Asterisk Users Mailing List - Non-Commercial Discussion ?mne: Re: SV: [Asterisk-Users] Queues

Well, yes and no. I tested that before and it causes a silent ring instead of a call rejection. I actually want to disable the entire feature. So the phone always rings unless you're actually on the phone. Thanks for the reply though! Regards, Jan ________________________________ Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Jerry

Hi, I have several Polycom 501 connected to asterisk. The phone has an ACD-login function that I'd like to use. But I can't find find much information about this. I've read a post on bugs@digium (http://bugs.digium.com/view.php?id=6119) about this function but I'm not really clear on if this is actually working or not? Has anyone actually used the Polycom ACD-login function

I don't quite follow you? There are USB headsets that don't require a soundcard at all. They have a built in soundcard which (I suppose) could be better than the crap they build into most laptops. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r El Flynn Skickat: den 24 maj 2006 10:17 Till:

Hi, This issue has been discussed probably a million times on every asterisk forum in the world and I have the same problem too. Another problem you would have with the agents is that when they make an outgoing call they are not regarded as "busy" by asterisk and it sends more calls to the agent if it has call waiting enabled. This behaviour is totally senseless since the whole purouse

>From what I understand it means that the *hardware* in your computer *acknowledges* the call as soon as it is recieved and then sends it to asterisk dialplan for processing. You would essentially need to put the delay before the call ever reaches asterisk. So this problem isn't asterisk related... if I've understood your question and the answer I found correctly. Regards, Jan

http://lists.digium.com/pipermail/asterisk-users/2005-September/125146.html -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com genom Brian J. Murrell Skickat: to 2006-02-02 20:14 Till: asterisk-users@lists.digium.com ?mne: [Asterisk-Users] delaying "answer" for a number of rings or anamount of time I want Asterisk to delay answering the POTS line via a

Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really was no solution (?). I'm not sure if this problem has occured now because of more load on the

Hi Chris, Yes the call was from PSTN and your solution worked great! I've read about SetCallerPres earlier but I didn't connect the dots this time. Thanks alot! :) Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Christoph F?rstaller Skickat: den 29 mars 2007 15:29 Till: Asterisk Users

Kernel is 2.6. zaptel and ztdummy load with Linux, so I can check in lsmod that they are loaded. They load without any problem, I've loaded them manually too. Zach A -----Original Message----- From: Dovid Bender [mailto:asteriskdigium@yahoo.com] Sent: Friday, March 10, 2006 6:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] how to check if

Hi, What USB headset would you recomend? We have some laptop soundcards that are really bad and I would be glad if you could share your experiences when changing to a USB headset instead of using the built in soundcard in your computer. Thanks! Regards, Jan

OK! I'm sorry about this. As I wrote earlier I'm totally lost... but I will try to explain the problem in steps bellow, ok. 1. I installed the rpm's for samba, Version 2.0.2a-ssl I think this is the version distributed with redhat linux 7.0 2. Then I changed the parameters in the /etc/samba/smb.conf file, and in this file I added the folowing parameters. [global] netbios name

Hi all, What are the best and free tools for remotely adding, removing users on Asterisk server and also for monitoring the status of the Asterisk server, like how many users are logged on etc. I need tools for which I don't have to pay. Thanks, Zeeshan A Zakaria www.acabling.com <http://www.acabling.com/> -------------- next part -------------- An HTML attachment was

Hi, Is it possible to keep asterisk from logging exactly everything? I can do the logger rotate and keep the files small enough, but I think it's unneccesary to log exactly all data. File grows by about 5 gb per month! Thanks! Regards, Jan

Hi, Is it possible to configure asterisk so it doesn't play MoH from certain phones? Regards, Jan