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Hello all, I''m getting stucked with debian and Xen 3.0.2 Problem is that during boot with xen kernel system said: mount: unknown filesystem ''devfs'' and then Unable to open initial console /dev/console And reboot... How can I solve? googled for some information but nothing usefull.. Thanks Stefano -- Ste Visit: http://www.stefanocislaghi.it/

Hi, does anyone have any idea whether VIA VT6102 "Rhine II" and Realtek RTL 8100 Ethernet chips are supported by FreeBSD? The hardware list for supported chips does list 'VT86C100A "Rhine II"' but according to VIA website VT86C100A is actually just "Rhine" and it is obviously not the same chip. (The 'VT3043 "Rhine I"' chip mentioned in

Recently I wrote about this problem, but it still exist and I can't dial my Xlite SIP Phone So here is the Notice Jul 5 17:14:07 NOTICE[65541]: chan_sip.c:6731 handle_request: Registration from 'Damian Minkov <sip:damian@10.1.1.2>' failed for '10.1.1.11' The * box(10.1.1.2) and the PC(10.1.1.11) on which is the XLite are in the same network Here is part from sip

So far, I''ve only found references to success under VMware. But has anyone managed it, with either the open source or commercial versions of Xen? I like Xen and have worked with it extensively for RHEL and CentOS, but it could save me a whole lot of hardware money if I can decommission some OpenServer hardware and store working OS''s under virtual images.

I added the playback line to my [macro-stdexten] context but when I dail an extension I don't get the "please hold while I try that extension" message. It just dials the extexsion. Do I have a syntax problem somewhere ? exten => 8005,1,Macro(stdexten,8005,Zap/2) exten => 8006,1,Macro(stdexten,8006,Sip/8006) [macro-stdexten] ; ; Standard extension macro: ; ${ARG1} -

When I have an ip hardphone username setup in my sip.conf : [109] type=friend username=ipphone9 secret=bla-la host=dynamic dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info defaultip=172.20.0.139 mailbox=109 ; Mailbox for message waiting indicator callerid=ipphone9 <109> callgroup=1 pickupgroup=1 and this user has a wrong password then calls are denied, but

Hi folks, I want to fix hole in my asterisk set up. I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN, Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go 'other' places. This senario works fine. Now the issue is someone else running a vocal or another SIP proxy can redirect his calls to my * as well. Those calls two will come through general

...ot of code fixes to make Asterisk work better on FreeBSD systems. He's also fixed the recursive mutex bug that has been a show- stopper for some time. That fix is not in CVS head yet, but will make it there soon. Rich, thank you! The award, a nice cup of tea on my veranda, waits for you here in Sollentuna, Sweden. (Yes, this is a very personal choice, since I'm using Asterisk on FreeBSD. A one man dictator-award-jury in action :-) * Improved language support --------------------------- Thanks to our new bug marshal, Fran Boon, aka Flavour, we now have even better support for various languages i...

I have a PolyCom Soundpoint 500 sip phone. I'm tring to get the phone registered on an asterisk box but am having no luck. I get the following errors 192.168.22.196 being the phone and 22.254 being the asterisk box.. Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_request: Registration from '"110" <sip:192.168.22.196@192.168.22.254>' failed for

...support Digium. Let's discuss this and find a solution! I am sure that with a community as large and vibrating as Asterisk.org, we will find a way to solve this before the spring. We have a long, dark winter with plenty of time to sit indoors and think. At least here in Sweden :-) Cheers from Sollentuna! /Olle PS: If you have topics that you want me to cover in Asterisk [weekly/monthly/quarterly] news, please feel free to drop me an e-mail or find me on the irc.