Does anyone know if Gigaset is for sale in the USA?Based on my assessment of phones and features, i would like to try the N300IP base along with C610H phones. I can only find the handsets on ebay, no retailers in USA.And I suspect they are using Europ..
all, Ive done a basic install of 10.1.2 to have a play with the new ConfBridge application and have noticed high latency when in a conference. Its to the order of 900ms or so which is just too much for a conference to work well. I can account for ab..
Jabra headsets work fine with Polycom. Fra: email@example.com [mailto:firstname.lastname@example.org] På vegne af Blake Burgess Sendt: 7. februar 2012 05:01 Til: email@example.com Emne: [asterisk-users] Head..
Since the recent update to the NAT configuration options and defaultsin chan_sip.so, I am interested in any SIP/NAT best practices advice.What Ive always done in the past is:Global: nat=noSIP handsets that are local: nat=noSIP handsets that are remo..
Ive been away from asterisk for a while since 1.4.16 and only installed 1.6 once to run a test… can someone recommend what the best version to install is and the recommended CPU/motherboard for an * box these days? Im just running about 20 hands..
Hello Id like to configure Asterisk so that… 1. I ring it from my cellphone with CID number displayed, just to notify Asterisk that I wish to make a call 2. Asterisk waits until I hang up, calls me back, and prompts me for the number I wish to c..
Ive been experiencing trouble with my DAHDI channels for some time and have finally decided to try and resolve the issue. Essentially, the problem I am having is that when a call comes in, and my DAHDI phones therefore ring, Asterisk thinks that ..
We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk 🙂 1) Is there a handset that will do this? 2) Is there a different (standa..
I have a client that has an Ubuntu x64 based Asterisk server with and ISDN Dahdi interface and about 25 SIP handsets. Everything was working fine in Asterisk 1.4 and now after migrating the config to Asterisk 184.108.40.206 I have one single issue that I c..