similar to: Brandable SIP SoftPhone (Windows) ?

Hi List, (Long time lecher, first time poster) Not sure if this is the best place to ask, but here goes.. I have a few test Guests on a Xen host, and I would like to start monitoring (and maybe restricting) the network traffic of the guests but not the Host. No I know this can be done with iptables, but I just wondered if anyone knew of a package or how-to to set this up with Xen on CentOS 5 ?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? - -- Gavin Spurgeon. AKA Da Geek -

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List, I have come across the above handset a few times in the UK, They are quite cheap over here (~?80) Not the best handset in the world but works well enough. I have been asked to setup a central config server for a large collection of these handsets. I know they can do Auto provisioning via FTP/HTTP/TFTP I have got an example of the generic

Do you have to enable encrypted passwords for smbpasswd to work? I have yet to enable encryption because I wanted to test everything first, but have enabled 'update encrypted = Yes' so that this will update the smbpasswd file. I also have 'unix password sync = Yes'. However, I cannot get smbpasswd to change the passwords, either by user or root. Any suggestions? Carey

Hi together, I try to find a softphone (freeware) solution for Windows 32, that works without problems ... Right now I use iaxcomm wich was best, of the ones I tried. But I have one problem with it. When I turn on qualify, it will not connect to the asterisk. This is also documented and normal behaveor. But if I turn of qualify, iaxcomm does not reconnect to the server, when the server got

Interesting offering, free from Twilio, this is php you install on your own server to build a brandable "VBX". Worth checking out! Listen to tomorrow for more about this and talk to lead engineer or Twilio CEO if you have any questions; sip:200901 at login.zipdx.com or Skype:vuc.me IRC: #vuc on Freenode.net or http://vuc.me/irc Info about VUC is htp://vuc.me Best, /r

Hi, Few subjects cause as many arguments as "which SIP client works best?" on IRC #asterisk, voip forums, and probably the -users mailing list. I have tried most of the SIP clients available in the last 5 years, both with Asterisk and other platforms such as OnSIP.com, IConnectHere.com, ZipDX.com and the venerable old FWD (in the days when that almost worked). Speaking of ZipDX, we

Hi, In the aftermath of Digium's and Counterpath's Bria for Asterisk announcement, we're happy to chat with Todd Carothers, Counterpath Product Manager today at 1 PM EST. For more info, http://vuc.me Join us on IRC #vuc on Freenode.net or use the web client at http://vuc.me/irc Call in starting at around 12 Noon EST: sip:200901 at login.zipdx.com Hear you there! /r

http://dovecot.org/releases/2.0/alpha/dovecot-2.0.alpha3.tar.gz http://dovecot.org/releases/2.0/alpha/dovecot-2.0.alpha3.tar.gz.sig Nothing horribly exciting since alpha2, but since I did other releases I thought I might as well release this one too. - Most of the stuff in master.conf is now built into binaries so the file itself is small. dovecot -n also doesn't show the defaults anymore.

Is there a way to have users of the samba server, but not add them by smbpasswd -a <UserID>? I want the samba server to be a domain member and the users to only authentic from the PDC. These are the steps that I have attempted: Users are in both the Windows domain and the UNIX NIS account 1. smbpasswd -j <Domain> -r PDC -U <admin> Joined the Domain 2. edited the

Hi List... Lets say I have been asked to do an Asterisk/Trixbox install for an environment where 2 companies 'live' in the same... building... Both companies have 2 incoming Analogue BT Lines and then have good old BT Phones plugged into them.. I have been asked to setup a VoIP system for one of the companies and the other company has showed an interest, There is one central Server

Which rpm has libstdc++.so.5? xten-xlite for linux says it needs this. Of course, there will be something else it will need after I get this....

Hello I need to hook up someone's remote PC onto our Asterisk server over the Net. There are firewalls on each side, so I figured it's time to give IAX a try, and see if it's less of a pain to use than SIP. And since IAX hardphones are pretty are, I guess I'll go softphone. Apparently, the two most well-known IAX and SIP clients for Windows are ZoIPer and X-Lite, respectively.

Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. It looks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126

asterisk-users-request@lists.digium.com wrote: > Date: Sun, 22 Apr 2007 19:38:04 +1000 > From: Rob Hillis <rob@hillis.dyndns.org> > Subject: Re: [asterisk-users] Softphone that supports central > provisioning? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <462B2CFC.50709@hillis.dyndns.org>

Mostly to debug/test BLF, is there a softphone or another app. which can subscribe to hints on Asterisk? Heck, it doesn't even need to be able to do calls :-) Leif

FYI:, I'm fairly new to samba (a couple of months now), and have experimented - unsuccessfully - with getting it to work with clearcase. Does anyone have a how-to, or a list of gotcha's for setting up clearcase with samba? Thanks in advance for any information you can send me. Ozzie,

Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? Reagrds; jonnyhashem --------------------------------- Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all my asterisk server, 2 sip client softphones are the same LAN asterisk ip address : 192.168.1.5 sip client 1 : 192.168.1.4 sip client 2 : 192.168.1.2 asterisk starts ok with sip setup the sip.conf [test] type=friend username=test secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes [1000] type=friend username=1000 secret=1000 host=dynamic context=cucku

Hi, I've been testing out the speex narrowband codec at low data rates (using linphone and Counterpath's Eyebeam). I'm finding that at data rates of ~25 kbps, the quality of the voice call is very poor. I know speex is supposed to work at much lower data rates (~2 kbps). Has anyone verified that speex will produce reasonable quality at low data rates? Are there any existing