similar to: "Random" all circuits busy now message

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

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

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, 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

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, I've set up FreeWorldDialup on my asterisk server but when I dial the service numbers, I get message '486 Busy Here '. When I dial any other number, it says 'All Circuits are busy now'. What is the problem with my settings. I've followed all the instructions step by step. Zeeshan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everybody After setting up trunk with FWD, all I get on my Asterisk box is message saying that all circuits are busy now, try your call later. Even 612 (time) says the same thing. Why is it that and how can I fix it. Zeeshan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am not sure why I am getting this message, I have an outbound route that goes to asterisk gateway1 then asterisk gateway2 When all lines on asterisk gateway1 are full, I get the message " all our circuits are busy now" then few second later, the phone rings, going to the second route! And the call can be established, how can I get rid of this message?? thanks --------------

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

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

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

I have downloaded and installed Asterisk@home and I have installed X-Lite on my Windows machine and I am able to connect it to the Asterisk server. I went ahead an created an account on Broadvoice today and followed the directions on http://voip-info.org/wiki-Asterisk+settings+Broadvoice and http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup but when ever I try and make a call from

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

Hello, I have a weird case, when some numbers dialed using a PRI, have an early media sounds instead of normal ringing. Few of the numbers are making Asterisk 1.6 (using Elastix 2) to report "all circuits are busy now". All of this numbers are cellular phones, but they constantly reporting the same thing. But most numbers even with such music, works well. What do you think I should

When using a PRI on my asterisk system, and when calling a busy number I often get an "All Circuits Are Busy" message. I've trouble shot it and made absolutely sure all circuits are not busy... but the number being called is in fact the issue. How do I get Asterisk to give me a busy signal instead of "All Circuits Are Busy" when calling a busy number? Is this something

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 =>

hello list, i have asterisk 11.15.0 and i have some trunks sip from my provider we have some ip phone astra 6731i each Ip-phone is configured with trunk and we call no ihave configured another trunk from the same provider in my asterisk i can call all numbers just the numbers are configured in thses ip phones. but when i configured the same trunk in x-lite i can call theses ip-phones without

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