similar to: hdlc + te410p + kernel 2.6.9 - anyone done this?

All, I'm having a heck of a time getting hdlc to work on kernel 2.6.11.9 .. I compiled hdlc, hdlc_gen, hdlc_cisco, hdlc_raw, into the kernel (note into, and not 'modules'). System comes up, I configured zaptel.conf span=1,0,0,esf,b8zs nethdlc=1-24 modprobe wct4xxp ztcfg sethdlc hdlc0 cisco ifconfig hdlc0 up All of this works fine, believe it or not. I have a T1 cross plugged

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

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

Dear All, we have installed a TE410P card on a Dell Poweredge 1750 running a slackware 10 with 2.4.26 Kernel. Then we have made two loops on the card and we have configurated all the 120 channels. Our goals was to perform some stess tests even if in this scenario we used the same box as generator and target. The stress test comprised to generate up to 60 calls at the same time by placing a file

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

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

I haven't seen much activity on the list concerning Digium's IAXy standalone FXS device. Are they worth $100 ? Do they integrate well with * as an extension's FXS ?

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

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

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 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:

--- 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

Hello everyone, We have recently turned up a new T1 from McLeod (Midwestern CLEC). It is configured like so: /etc/zaptel.conf: loadzone=us defaultzone=us span=1,0,0,esf,b8zs ;(also tried 1,1,0,esf,b8zs) bchan=13-23 nethdlc=1-12 dchan=24 /etc/zapata.conf: switchtype=national context=pri-in signalling=pri_cpe group=1 channel => 13-23 I can get hdlc0 (and pvc0) up just fine after the

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

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

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

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

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 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