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<!DOCTYPE html> <html lang="en-US"> <head> <title>OpenWrt Forum Archive</title> <meta charset="UTF-8"> <meta http-equiv="X-UA-Compatible" content="IE=edge"> <meta name="viewport" content="width=device-width, initial-scale=1.0"> <link rel="stylesheet" href="assets/css/common.css"> </head> <body> <div class="container"> <header class="main-header"> <h1 class="logo"><a href="./"><img src="assets/img/logo.png" width="376" height="88" alt="OpenWrt Forum Archive"></a></h1> </header> <aside> <p>This is a read-only archive of the old OpenWrt forum. The current OpenWrt forum resides at <a href="https://forum.openwrt.org/">https://forum.openwrt.org/</a>.</p> <p class="minor">In May 2018, the OpenWrt forum suffered a total data loss. This archive is an effort to restore and make available as much content as possible. Content may be missing or not representing the latest edited version.</p> </aside> <main> <header> <h1><span class="minor">Topic:</span> openwrt - asterisk performance</h1> </header> <div class="notice minor"> <p> The content of this topic has been archived on 25 Mar 2018. There are no obvious gaps in this topic, but there may still be some posts missing at the end. </p> </div> <div class="pagination"><div class="pagination-number">Page 1 of 1</div><nav><ul><li class="pagination-current"><span>1</span></li></ul></nav></div> <article class="post" id="p79855"> <div class="post-metadata"> <div class="post-num">Post #1</div> <div class="post-author">brcisna</div> <div class="post-datetime"> 19 Jan 2009, 05:39 </div> </div> <div class="post-content content"> <p>Hello List,</p><p>Could anyone here post their real world results using openwrt and asterisk . I have both an wrt54g v.2 and wrtgs. Just curiuos if the audio is actually decent etc. I know asterisk can be touchy to get working for people with lots of experience with it for many installs,from what I heard. Also who are you using for a voip provider,and are you happy with them? Do you have voicemail,and call forwarding setup,etc.</p><p>Thanks,<br />Barry</p> </div> </article> <article class="post" id="p79856"> <div class="post-metadata"> <div class="post-num">Post #2</div> <div class="post-author">frogonwheels</div> <div class="post-datetime"> 19 Jan 2009, 05:54 </div> </div> <div class="post-content content"> <p>Hi Barry,</p><p>I've been working with xMff to get asterisk working with a Luci front end. I use my Asterisk at home on my Asus WL500GP - both my wife and I work from home.</p><p>I've been able to cut the modules loaded right down - and have also broken up asterisk into many modules (about 80 iirc). <br />The modularised version and the UCI configuration generator is a WIP and is in the Luci Devel SVN (under asterisk-xip). The LUCI front end is very much a WIP.</p><p>I am running Asterisk 1.4 with voicemail, MeetMe and some other stuff. <br />I only use ONE encoding (ulaw) to avoid any transcoding (apart from sound files in gsm)</p><p>I haven't stretched the system to breaking or anything, but I have been able to get <br />2 External Calls + 1 Internal on a MeetME Conference call at the same time as answering another incoming call.</p><p>The audio is fine 95% of the time. You get the odd glitch and outage - but it's hard to know what the cause is.</p><p>I use my ISP as my provider... which probably is a good thing .. means the servers are only a hop or 2 away from my ADSL2+ connection.</p><p>I can transfer calls from my phone - but have no forwarding setup. I've transferred external calls to another external number without glitch.</p><p>hope this helps</p> </div> </article> <article class="post" id="p79876"> <div class="post-metadata"> <div class="post-num">Post #3</div> <div class="post-author">brcisna</div> <div class="post-datetime"> 19 Jan 2009, 14:20 </div> </div> <div class="post-content content"> <p>frogonwheels,</p><p>Thanks for the info. Could you elaborate on what version of OpernWRT you are running?. Is your ASUS router running on strictly the flash memory or is asterisk actually residing on a usb stick Also a biggie, what brand/model of phones did you decide on?It seems there ends up being horror stories on about any given brand/model,if you search enough. I m not exactly sure what you mean "wip".... I haven't been around openwrt long enough I guess:). What kind of actual bandwidth are you getting from your isp up & down? I switched over to cable ( from a very reliable dsl/phone contract ),,, just to try something different and some days my down speed is only 500kb's!,,,,which is plenty adequate for a couple voice calls,( once i get asterisk going),,but that's cutting it pretty thin.<br />Let me know were to get the asterisk modules you are speaking of ,if you would:).</p><p>Thanks,<br />Barry</p> </div> </article> <article class="post" id="p79928"> <div class="post-metadata"> <div class="post-num">Post #4</div> <div class="post-author">frogonwheels</div> <div class="post-datetime"> 19 Jan 2009, 23:01 </div> </div> <div class="post-content content"> <p>Barry,</p><p>WIP = Work In Progress</p><p>I'm running on a svn trunk build of OpenWRT - possibly a month or so old now. I'm running asterisk off a USB stick - which helps for things like voicemail. I could possibly run the modularised version off the flash - there's much, much that I don't need (including a bunch of sound files).</p><p>I'm actually, at the moment, just using a linksys PAP2T with standard phones, my Nokia E65 mobile as a wireless SIP client, and a couple of zoiper instances on a windoze box.<br />Something like:<br />Download Speed: 8000kbps (1000.0 KB/sec transfer rate)<br />Upload Speed: 821kbps (102.6 KB/sec transfer rate)</p><p>Currently the makefile that breaks asterisk up into modules still needs some tinkering before I request it be pushed into trunk - so those modules aren't available yet.</p> </div> </article> <article class="post" id="p79942"> <div class="post-metadata"> <div class="post-num">Post #5</div> <div class="post-author">mazilo</div> <div class="post-datetime"> 20 Jan 2009, 05:08 </div> </div> <div class="post-content content"> <p>I have asterisk-1.6.x built into my own OpenWRT firmware from SVN trunk for my FON2100 WiFi router. I use G729 as default CoDec with G729 sound files. My FON2100 is connected to my main NAT/Firewall router with a <strong>NAT=yes</strong> and <strong>canreinvite=no</strong>.. As such, my asterisk will stay in the audio path and can't redirect the RTP media stream (audio) to go directly from the caller to the callee. Whenever there is an I/O activities on my FON2100, i.e. reading the Flash space (mtdblockd process), this will create some hick-ups (or temporarily losing audio signals). Other than this, my asterisk has been doing great.</p> </div> </article> <article class="post" id="p80001"> <div class="post-metadata"> <div class="post-num">Post #6</div> <div class="post-author">marca56</div> <div class="post-datetime"> 20 Jan 2009, 21:39 </div> </div> <div class="post-content content"> <p>Mazi:</p><p>What made you choose 1.6 over 1.4?</p><p>Which modules are you loading (e.g., meetme, voicemail, etc.)?</p><p>What are the file sizes of the apps and modules combined?</p><p>Thanks.</p><p>marc.</p> <p class="post-edited">(Last edited by <strong>marca56</strong> on 20 Jan 2009, 21:41)</p> </div> </article> <div class="notice minor"> <p>The discussion might have continued from here.</p> </div> <div class="pagination"><div class="pagination-number">Page 1 of 1</div><nav><ul><li class="pagination-current"><span>1</span></li></ul></nav></div> </main> </div> <!-- Created in a hurry and not indicative of usual code quality. Here's a number: 0 --> </body> </html>