similar to: Duplicating My CentOS Install For Other Other Systems

I seem to remember this being addressed before, but I can't find the howto anywhere. I've got a friend heading to another country with VERY limited bandwidth. He'd like me to update the 4.4 DVD to include all of the updated RPMS from updated. Where can I find the scripts to update the meta data on the RPMS and create a new bootable DVD? Thanks! Ben

Hello, Does anyone have an up-to-date page describing, step by step, how to make a customized CentOS5 DVD? I noticed that CentOS5 already comes with ~240MB of updates. So for starters, I'd like to create a new DVD with all the current updates. (And I have other custom scripts I need to install on top of that). I've googled around and tried various suggestions on the net:

I recently purchased a Mediatrix 1124 from an auction of a company that went out of business. It came with nothing other than the unit itself. In digging thru the Mediatrix web site, and various google searches, it looks like it only supports SNMP setup, and only with their software (or the correct MIB). However, Mediatrix doesn't appear to let you download said software or MIB from

Hi. there is a way to MARK udp VOIP (SIP) traffic, in order to put in a highest prio class ? Traffic flow seems start on udp 5060 port, but next both server and client seems jump to a random(?) port. I can''t use CONNMARK because is udp traffic. I only see a pattern for L7 patch in order to SIP traffic identification , but I run 2.4 kernel series . When you patch 2.4 kernel with

I have four Polycom 330 phones connected to an asterisk system. There are other VoIP phones connected too. All of the extensions are four digits beginning with 11. From any of the phones, except the Polycom, picking up the handset to call extension 1103 for example works fine. With the Polycom 330, as I press the second 1 of 1103 it stops taking input and gives me an error. I tried

Hi all, I am running centos 5.1 and I wish to change the domain name and dnsdomainname of my PC. currently the settings are-- $ hostname sipx.com $ hostname --fqdn sipx.com $ domainname (none) $ dnsdomainname com I have searched in the net for tips but everywhere only the hostname change is provided. I need to change/set the domain name and the dnsdomain name on my pc to sipx.com and this

Hello, I just had a meeting about a pilot project going on in our University, The project manager has done some research in the past year and concluded that Asterisk can not scale well to large user base like 10,000 users, thus Asterisk is not fit for large University environment. The project manager instead choosed sipX and said it scales well for large user base. I had an Asterisk running

Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>

How do you configure your Polycom phones? Is it enough to configure one line appearance? Or is there a way to configure a roll over? Chris Mason US Number: (646)722-0001 US Fax (815)301-9759 Skype: netconcepts

Hi all, I want to deploy a SIP Proxy but I just don't know which one to choose. Researching in the Internet I found the following ones: * SIP Express Router <http://www.voip-info.org/wiki/view/SIP+Express+Router>: SER is used by many SIP providers standalone or in conjunction with Asterisk * Vovida.org <http://www.voip-info.org/wiki/view/Vovida.org> * sipX

Maybe I just woke up too early today. I have SJPhone and X-Lite working perfectly but I cannot for the life of me get sipXphone working properly with Asterisk. Its probably something stupid on my part, but does anyone have a quick setup sheet for it? -Kerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have installed astriesk 6 and am able to make calls using sip x-lite. Its working as I expected. Now I want to make call from sipx-lite to PSTN using asterisk. can any please suggest me which Hardware card that I can buy? and use ( pl give me all ths list of cards.... which are good.). 2) what is that I need to do after buying the card to make it talk to the real world PSTN network?

Hey there. I've been working on a project to integrate Asterisk with Exchange Unified Messaging via sipX using large parts borrowed from: http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html ... and everything works surprisingly well. The one problem I have is MWI, or a lack thereof. Exchange 2007 doesn't support MWI of any kind (!), so I've been looking into

Has anyone compared * to sipXpbx? From a cursory look, this open source version of PingTel's PBX has many features that make it more suitable as a replacement for a traditional PBX, including the ability for users to tell if a phone/trunk is in use. What I am trying to figure out is what I'd give up using sipX instead of * (and vice versa). /carmi

Support for SLES9 seems to be pretty much lacking at the moment. The OCFS2 site [1] states OCFS2 is supported with SP2+. There actually does not exist a SP2+ for SLES9 (or should this rather read "will be supported in SP3" instead?). Latest RPMs that come with SLES9 SP2 are 0.99.14 which has critical known bugs and is unusable. Anybody got to compile current source on SLES9 x86_64?

Hi, I'm putting together a fax solution for my company that utilizes T.38. I wanted to throw out my plan and get some feedback if anyone is doing something similar or sees a blatant problem with it. We're currently rolling out SPA-942 phones for the standard desk phone with vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls for satellite offices are handled by

Thomas B. Ruecker wrote: > Michael Grigoni wrote: > >>Greetings: >> >>It would be nice if Icecast supported RTSP; > > It probably never will > >>however I would >>appreciate any suggestions for a small RTSP/RTP solution to >>encode 8kHz mono audio in GSM or ADPCM and service multiple >>unicast client connections. > > why not use

Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat' soft-key to work? When you change the status in this way, the phone does not send any communication to Asterisk, and it seems to have no effect in incoming calls. So... what's it for? Doug

Hi: I'm using speex to perform echo cancellation in Windows. I'm aware of the problem about out of sync clocks in record and play sample rates in usual sound cards . In order to have an idea of how good is my echo cancelation working I would like to know if there is any #define thing i can pass to speex_echo_ctl to get the actual level of echo cancellation. If not, how can i extract that

This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards,