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Greetings list, Thanks to all who replied to my thread a few days ago "SIP devices with packet loss tolerance". One of the suggestions that came out of that thread was to replace routers at users' premises with ones that support QoS. I've used m0n0wall's QoS in the past with reasonable success, but it's quite a bulky and complex setup for deploying to remote sites which

Hello List! I wanted to share to everyone the following compatible connectivity products that my company installed in our Asterisk based soft switch. I already sent these to the Asterisk.org site many days ago but for some reason they still have to post it. 1. Sangoma A101 single port E1/T1/PRI Card 2. Sangoma A102 dual port E1/T1/PRI Card 3. Sangoma A104 quad port E1/T1/PRI Card 4. Sangoma

I'm having problem connecting to my isp with my sangoma s518. Everything seems to run smooth and wanrouter are training to connect....but forever.. Now, to test the connection i mounted up my standard nic and my old dsl router, to establish a standard basic pppoe connection..... and I can't with centos. I follow this standard setup: http://www.redha...

(originally filed as a bug in Fedora's bugzilla, see https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=171933) Greetings, Using Fedora Core 4 on a Dell PE 420sc. Malfunctioning kernel is smp-2.6.13-1.1532_FC4. Properly functioning kernels included smp-2.6.12-1.1456_FC4. Network performance is extremely poor when using bridged network interfaces. When not using brctl, the interface

...gh the wanpipe install instructions and configured it, now I can run [root@altpbx asterisk]# wanrouter hwprobe ------------------------------- | Wanpipe Hardware Probe Info | ------------------------------- 1 . AFT-A101u : SLOT=1 : BUS=1 : IRQ=209 : CPU=A : PORT=PRI Card Cnt: S508=0 S514X=0 S518=0 A101-2=1 A104=0 A300=0 So I know the card is there OK. My /etc/zaptel.conf looks like: span=1,1,0,esf,b8zs loadzone = us defaultzone=us fxsls=1-12 I am only trying to get half to load for now to make it simple. [root@altpbx asterisk]# ztcfg Notice: Configuration file is /etc/zaptel.conf l...

Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards

I upgraded from Tormenta2 to Sangoma A101. I followed the instructions, and installation was successful. zttool, ztcfg, all show card is installed properly. I copied the parameters from my old working zaptel.conf, zapata.conf and zapata-auto.conf. Verified on Sangoma website that these files are correct. Also configured wanpipe1.conf. But doing all this didn't start the PRI channels. It says

I have a very annoying problem that we hear on our end, but the other party doesn't hear. There are random beeps and echo type noises that occur. They are present during voicemails, and present on my end during calls. Is anyone experiencing the same deal? I have asked this a number of ways on the list, and never get a response... Thank you. Sean Garland Mount Shasta, CA

My router (1605R) currently does not support QoS. Is there any open source software available so that I can set one up before the router? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/38b6e3f2/attachment.htm

>Message: 23 >Date: Fri, 13 Jul 2007 22:11:59 -0400 >From: Dan Halbert <halbert at everyzing.com> >Subject: Re: [CentOS] Newbie ADSL configuration, ppp0 can't activate & > config not found >To: CentOS mailing list <centos at centos.org> >Message-ID: <469830EF.3080601 at everyzing.com> mailing-lists at computer2.com wrote: <snip>

I am currently experiencing intermittent silences with my asterisk system. The symptoms are as follows: * Both for incoming and outgoing calls, I (and other users) occasionally experience a brief period of silence. * The silence lasts anywhere from 3 to 10 seconds. * It is not due to silence suppression, because the silences generally occur in the middle of sentences. * Silences occur at

Hi, We have Asterisk, running on a P4 box running Suse 9.1, making calls using IAX through SimpleTelecom and Nufone. What we are looking for is toll quality voice. The problem is that voice over calls routed through SimpleTelecom and nNufone occassionally breaks. We also have a digium card and the calls over the digium card using the Zaptel Interface have a very good quality. We

As seen on my post at: http://www.sveasoft.com/modules/phpBB2/viewtopic.php?p=28112#28112 This works very well... It does NOT work with stable 4.0! sveasoft will be issuing a bug fix for this (4.1) in the near future. Final Rev of working script w/ asterisk support I'm not going to run alchemy on production machines until it is stablish. Remember to set your uplink properly and to set