similar to: Any good guides for installing Asterisk on Embedded systems like Alix boards?

Hi, I'm looking for a good network device that does bandwidth management. It can be integrated in a router or stand-alone, but must be SIP-friendly. I`ve tried the DIR-655 (latest firmware is SIP-hostile, and the latest hardware revisions can't downgrade to the version that worked well) and the DI-724GU (SIP-friendly, but bandwidth management is automated and not configurable

Has anybody tried running Xen on an ALIX 2D3 board? http://www.pcengines.ch/alix2d3.htm If not, has anybody ran Xen on a seriously underpowered PC? The reason why I love these boards so much is that they run at 5W! Of course, I would only run Xen on these for testing purposes. It would just be nice to have a very low powered system that I can just "blow away" VMs when I

Hi All, I am trying to upgrade an old Asterisk installation to 1.4.2 (it's currently running CVS-08/02/04-15:15:26) but have hit a couple of problems. The first was easily fixed. I got "storage size of sin isn't known" errors whilst compiling streamplayer.c, but after seeing http://bugs.digium.com/view.php?id=4908#32012 I manually added "#include <netinet/in.h>"

The AstLinux Team has released 1.2.0. All current users are encouraged to upgrade as this release addresses the bash "ShellShock" bug. New in 1.2.0: * New Linux Kernel 3.2.x * "igb" ethernet driver for Intel Atom C2000 * Enable AES-NI support * New "sip-user-agent" firewall plugin * New versions of Asterisk 11 and 1.8 * Bash "ShellShock" security fixes A

Hello Since TDM cards are known for being particular when it comes to motherboards (PCI 2.2, etc.), I was wondering if there is a utility that can check that the Zaptel driver works OK and can tell if the TDM card is compatible? That way, if an FXO module is not reporting an incoming call, we'd know it's because of the Zaptel driver, and not something elsewhere. Are "dmesg",

Hi there, Here is my dilema. I have a new small business customer that wants me to put in a VoIP phone system for them. Based on their requirements, I have determined that it needs to be a "set it and forget it" type of thing like a lot of small business proprietary systems. At the same time they would like to be able to do minor dial plan changes themselves so I have determine

The AstLinux Team would like to announce that the 0.7.0 version of AstLinux is available for download. There have been many significant updates in this release including updating to the latest Asterisk Release (1.4.29), moving to DAHDI (2.2.0.2) along with several other system updates. For a complete list of changes, read the changelog available on the download page:

Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing "sip set debug peer PROVIDER": Sending to 123.123.123.123 : 5060 (no NAT) ^^^^ That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see

Hi, I'm relatively new to Asterisk, and I'm looking to build a tiny system for home use. For context, at home I've got a line from Vonage (last I heard, they won't give out SIP credentials and let you use random hardware/software), which comes out as an analog line with a dialtone, and I'll be treating that as if it were a regular PSTN connection. I also work from home,

Hi Everyone, Has anyone got any opensource speech recognition software to work with Asterisk? Please only list WORKING ones. Not the "theoretically" should work ones! Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100821/4d11d6c0/attachment.htm

Hi Everyone, I have setup an OpenVPN tunnel between Server A (running Asterisk) and Server B suppling it's SIP Phones with DHCP pool of IPs. So, the tunnel is established nicely and everyone can ping others. "sip show peers" shows the local subnet of the SIP Phones registered (192.168.100.0/24 ). But there is the old bad one-way audio. Calls also drop after few seconds. In the SIP

Hello list, anyone here using Asterisk together with HTB for queing incoming and outgoing packets ? I've tried to subscribe myself to the Mailinglist of the Linux Advanced Routing & Traffic Control project, but I get no confirmation. This list seems dead. It seems my test case with HTB is not giving any noticeable results. Can I ask questions on this mailinglist ? Perhaps you can

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton

I've just posted this to another list where we were talking about the same old issues we've been plagues with recently - I'd already posted some iptables rules, but added more to it for this... This script probably isn't compatable with anything else, but I don't run anything else. It's also designed to act on the incoming interface, not to run in a router, but

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1"> </head> <body text="#000000" bgcolor="#ffffff"> <font size="+1">Does anyone have links to the most recent audiocodes

Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/92c7f946/attachment.htm

How can we set the CDR Master file to rollover at say 30 Meg and create a new one -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/2e98385f/attachment.htm

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm