search for: pikkarainen

Hi, in the Snom FAQ I found the following information: After staring up, the phone tries the URL given in the "Setting URL" of the phone. ... BTW this setting can also be set via DHCP. .... option tftp-server-name "http://192.168.0.9/snom200{mac}.htm" The documents used: FAQ-04-06-14-sf.pdf "Setting up DHCP for snom phones" FAQ-04-03-24-sf.pdf "How can I

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured

23.09.2011 11:31, Janne Pikkarainen ?????: Thank you for reply, BTW, other webserver has almost the same bonnie results (10283ms and 5884ms) on ext3 partition with 45GB of data (1.5 millions of files)?! Hardware and RAID5(also hardware) are the same: HP Proliant DL380 G4 with SmartArray 6i controller (as I see it comes with 128MB B...

Hello All I am looking for a PC based switchboard application. Cisco CallManager has a web attendant console that allows you to use the PC to transfer calls and the like and I was wondering if there was a similar program compatible with *. Thank you in advance Keith D'Atrio -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've had a search through the archives and didn't find much. Is anyone using the Monitor application? I have it working but there is a really big drawback. The files are always called the same thing, which means if I make 2 calls one after the other the first recording is lost. I half expected Monitor to use something like ZAP-2-1-<yyyymmddhhmmss>-in/out.wav for it's

Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying

Some of you have got Wifi600 wireless SIP phone working with Asterisk. Specially John Todd ( nice review ). My phones register ok. They can also receive calls from other phones. But for some reason I can't make them call out ( anybody, ie. SIP or PSTN ). This seems to be due to the phone not understanding what it should do when it receives 'Proxy Authentication Required'. In my case

...g). Server is a dual P3 Xeon 700 MHz / 1 GB RAM / 200 GB (hardware) RAID-5 SCSI. Distribution used is a fresh Red Hat 7.3 install, updated with up2date whenever necessary. Kernel version: Linux xxx.xxxxxxxxx.xx 2.4.18-4smp #1 SMP Thu May 2 18:32:34 EDT 2002 i686 unknown Best regards, Janne Pikkarainen