similar to: Asterisk snapshots?

--- hgaillac-sip@yahoo.fr a ?crit : > hello, > > How asterisk could support res_snmp even this module > don't help to monitor all asterisk features? > > monitoring asterisk with snmp would be a good > thing. > Which solution ? > > Harry > --- Kristian Kielhofner <kris@krisk.org> a ?crit : > > > hgaillac-sip@yahoo.fr wrote: > > > I

While not strictly Asterisk related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html -- Kristian Kielhofner

Hello everyone, Even though a lot of it was a bit last minute, several of us from the commnunity made it to Baltimore to help Digium with their booth at ISPCon. It was a great time. Gregory Boehnlein, Brian Capouch, Christian Savinovich, Kristian Kielhofner (me), and John Todd (not pictured) were there (as well as others), and some pictures were taken (the up close ones of me were very

Does anyone have a good list of links to the various presentations at Astricon, specifically one including a link to the performance analysis by those guys from Belgium? I would love to get a closer look at their graphs because it was impossible to read them, and I was pretty close to the front! -- Kristian Kielhofner

Happy New Year in advance by a few ticks for the northern hemisphere. Here's the first topic and guest for 2009: In any voice path there are several potential sources of quality problems, ranging from echo to voice dropouts and everything in between. With VoIP systems the potential for quality problems increases dramatically, often times making it very difficult to identify the source of

I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only "format=g729" in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the

Hello everyone, As I promised at eTel last week, I have finished up work on my "Asterisk Native Sounds" project. Here's a little diddy from astlinux.org: ----------------------------------- Asterisk Native Sounds are a collection of audio prompts for Asterisk. They will improve quality, reduce CPU usage, reduce latency, and (in some cases) eliminate the need for G729

Hello everyone, I just ordered one of these: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp Just over $110 with shipping but they are expecting the price to come down quite a bit: - 1.2Ghz ARM5 - 512MB RAM - Multiple flash storage options - Gigabit ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I

Hey everyone, A while back I worked on a project to measure call quality. I've finally gotten around to releasing it and I'm calling it recqual (Real Call Quality). There isn't much to it and it should be considered alpha quality. I'm hoping some of the bright minds on the list can help me out with it. I'll include the intro text from the README in the tarball: ----

Hello everyone, I was just looking over some logs, and it appears that there have been less than 3,000 downloads for my native Asterisk sounds packages (all formats combined). What gives ;)? In my humble opinion, EVERYONE (unless you have your own in a different voice/language) that uses Asterisk should be using these prompts. How about a direct link this time:

This was so interesting I had to move it to its own thread! Is anyone using this script? How does it perform compared to the older WonderShaper script? -M- ================== Thanks Kristian I will checkout the new script and see how it goes! Jonn -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at

Hello all, This might be one for Digium, but I would like to see some type of Wiki that people would have to wade through before they would get the information on how to subscribe to the list. This wiki should cover most of the basic stuff that gets asked over and over again just to help reduce the amount of repetition that most of you have probably noticed takes place here. I understand

Hello list, I have several SPA-2000's and 3000's scattered about the Internet (all behind NATs). Because I do not qualify as an ITSP, Sipura will not license their "Sipura Profile Compiler" so that I can have the units remote upgrade, remote re-configure, etc (via TFTP or HTTP). This is extremely annoying. Right now if I have to make a config change to any of these

Sorry about this, but do any users have more detailed iax.cc information? Will they do trunking? What are the hosts that I will be logging into? I want to make sure that they will work well for me, and I would like to do some traceroutes to make sure that they are close! Thanks. -- Kristian Kielhofner

I've been spending some time looking at some of the significant changes Apple has made to Facetime in iOS 7. I'm far from an Apple fanboy but some of them are pretty interesting: - multiplexing everything over a single UDP port - deflate compression with SIP - various /slight/ protocol violations ;) More here: http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html As

Hello, After checking out CVS HEAD from yesterday (for those new PLC/Jitterbuffer patches), I was affected by bug 3795 with my Polycom IP600's. After seing it resolved as of this morning (thanks Mark), I decided to try again... I can answer incoming calls. No problem there. Putting calls on hold, however, results in my Polycom IP600 indicating the call on hold, but the caller does

Hi Kristian, Anywhere I can read about this Soekris/AstLinux project? ... Regards, Hans -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Kristian Kielhofner Sent: Thursday, February 24, 2005 6:02 AM To: jim@vanmeggelen.ca; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

Hello everyone, I've had the time to examine some specific serial/parallel forking scenarios with Asterisk lately. Looking at chan_sip it appears that anytime Asterisk wants to tear down a call before it's brought up, it sends a 603 Declined: } else { /* Incoming call, not up */ const char *res;

who can tell me how to do TDM over enthernet ? pc a connect pc b only use TDM card? thank you John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?23? 11:47 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5, Issue 336 Send Asterisk-Users mailing list

Hello, I am just getting started with res_config and ODBC. I have MySQL all setup and am filling it with my data. Everything seems very straight forward. One thing catches me so far: 1) How are register lines in sip.conf and iax.conf represented? i.e. register=> username:password@fwd.pulver.com/700 insert into ast_config (filename,category,var_name,var_val)