similar to: Getting started with Xen

Hi List, Yesterday night after a power off due to a faulty UPS my asterisk doesn't want to start anymore. Here is what I get on the CLI: Asterisk Ready. *CLI> Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk uncleanly ending (0). I use 1.2.7 I think on a debian sarge and cdr_pgsql too. Any ideas?

Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? Thanks, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Hey guys. If my Asterisk box connects to the PSTN using SIP and IP over ethernet and doesn't require any authentication, what sort of a trunk would need to be created? Thanks, -- Nick e: nick.hoffman@altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality

Hello all, Just a question, on asterisk box : I looking on the web , for asterisk at large , and 'asterisk future of telephonie' ... If we would like to change our OLD PABX 600 phone with 4 E1, to install a asterisk with full ip phone in SIP, Could we use 1 Box for asterisk with voicemail, zap channels and some agi script ? thanks Fabrice

RAD appear to have bucketloads of products which bridge between various interfaces (E1, BRI, POTS) and their own TDMoIP protocol. The attractive thing about them for me is their availability in Australia. The voip wiki says not much about it (http://www.voip-info.org/wiki/view/TDMoIP), and certainly nothing about if there is any way to get Asterisk to talk TDMoIP. Despite the name, TDMoIP

Hi, I'd like to use the convenience of apt packaging, but debian sarge's default asterisk is something like 1.0.7. Are there any apt repositories which provide newer versions of the software? Thanks! -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Hi List, I was wondering if anybody had tried running Asterisk inside virtualization software such as Xen. Are there known problems doing it? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Hi, I need to have some SIP extentions on remote places where the latency from my asterisk box with public ip is 1~1.5 seconds. What codec will work fine on this sceneary? I'm planning to use iLBC, is a good choice? Regards, Guillermo.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I read in a message of december that packages of Xen 3.0.4 are expected in experimental by the end of Jan. Actually I didn't find anything in experimental. Since all my tries to create a kernel with xen 3.0.4 failed miserably with compiler errors. I had success only with the utilities packages, but I'm not even sure that they work. Is there

I am using the "tha" Xen packages from http://packages.debianbase.de/sarge/amd64/xen3/ However, I noticed that some new packages have been uploaded into the NEW queue. What I am wondering is if/when/where the Sarge packages will be made available. If there is no current plan to make them available, I would be happy to help in any way that I can. -Roberto -- Roberto C. Sanchez

Can anyone point me to a reference or sample config for bypassing a nailed up (point to point) t1 between two PBXs with asterisk and a pair of t1 cards? Right now I have 2 Nortel norstars connected to each other via a leased line t1. I also have a solid 10mbps low latency microwave link between the 2 sites. My goal is to run an asterisk box at each end with a t1 card and Ethernet card to

Hi, understand that the bandwidth utilized for each call is dependent on the codec used, wonder if Asterisk can monitor the total bandwidth utilized and restrict/reject new calls when the resource is insufficient to support them reliably? Regards Andy Tan -- Andy Tan andytan@fastmail.fm -- http://www.fastmail.fm - Does exactly what it says on the tin

Hi List, I am working on least cost routing code on the moment, and I am stumbling on a problem. Say you have provider A having: Prefix XXX 0.10 Prefix XXXYYY 0.20 And provider B having Prefix XXX 0.15 You're stuck, because you cannot decide if provider B's "XXX" prefix also covers XXXYYY numbers or not. If it doesn't, it would be a waste

Hi List, I am looking for a 1 FXO analog termination device, other than the obvious PC + FXO card, and which can achieve decent call quality. The SPA-3000 seems an option... have you got any other ideas? Cheers, Jean-Michel.

Hi All, I have a SIP provider that tells me that my RTP stream uses a "20bytes payload in the g729 coded data". And they would like that we change this to 30bytes (3 frames). But maybe I'm wrong but isn't a certain payload just a standard for a codec ? And if I'm wrong, how can I change the payload for my g729 calls in Asterisk. Greetings, Attilla

You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN

Slightly OT, but I think this is of possible interest to many of you, I need to get a UPS for my asterisk box. They are rated in VA but I can't quite figure out how that converts to real life. I have a PIII-800 box with two X100P and one TDM400P plus graphics adapter, an IDE hard drive etc. Will a small 400VA box be enough for this? tia

Hello, Could anybody tell me if there is a Linux distribution (or Kernel version) that works better with Asterisk. I am newbie and I don't know if there is a preferred Linux/kernel version for Asterisk. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040904/8798ff25/attachment.htm

Need help...I need to install a card to terminate 7 lines...I have not order the phone lines yet...I can either do analog lines 1FBs or order a fractional T1...any suggestions on what hardware would be easier to install and configure...also if I went with a T1...do I need an external CSU/DSU or anything or does it just plug into the T1 card...thanks.. -----Original Message----- From:

Greetings, We are currently testing a Sipura SPA-3000 as a gateway from our Asterisk system to a PSTN line for 911 access. We have a number of locations and want to place an SPA-3000 in each, connected to a PSTN line that will provide the correct ANI/ALI information to 911 for each location. It all works great, except for a reasonably significant (4 seconds) delay between when the SPA-3000