similar to: SV: BAD/GOOD Echo Cancel

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

What kind of help do you need then? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Zach A Skickat: den 6 februari 2006 18:31 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' ?mne: RE: [Asterisk-Users] Help on queues There is no good help on wiki and voip-info.org, I've

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:

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

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

>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

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

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

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, 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

Hi, I just read a pressrelease from VON that Digium will soon be releaseing a couple of new cards. What got me interested was: "The TE420P and TE415P support 128ms of G.168 (2002)-compliant echo cancellation across their entire 128 channels." Does anyone know when thease will be released and what they will cost when released? Thanks!

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

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

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

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

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

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, How do I read (make sense of) the timestamp in the queue_log? I'm probably just slow but I don't understand it. Thanks! Regards, Jan

Hi, I have a problem with setting outgoing caller id to "nothing" (secret) on our Wildcard TE405P connected to a swedish euroisdn line. Caller ID seems to work fine when connecting the same line to a Ericsson PBX - so something must be wrong in my settings, but I don't know what. I've tried: exten => _*70X.,1,Set(CALLERID(name)="") exten =>