similar to: Linux Bridge + QoS Shaper HOWTO available

Hello all, We've discovered that VoIP (IAX2) + Citrix + Video is pegging the measly CPU on the Netopia router our ISP provided. We've got 3Mb/3Mb and will increase to 4/4 next year. The Netopia simply breaks out our WAN IPs, and we've got a switch hooked up to it on the inside (Actually I've got a QoS box in-between). ------------- | Internet | | on Cat5 | -------------

I''ve created a pretty complete HOWTO on creating a Linux Bridge (using Fedora) to shape LAN <--> WAN traffic. It includes installation instructions, a script to configure the bridge (which you install as a service), and 2 scripts to configure the network interfaces using traffic control. http://www.burnpc.com/website.nsf/all/3a64a6369757819686256f960068ad75!OpenDocument If

Hi, I'm helping out with a political campaign and would like to use asterisk to blast out about 200,000 calls with a short message from the candidate. Provider: I'm thinking voipjet may be a good solution? Hardware setup: I will have access to several T-1 lines so I would just want to set up the dialers to limit the number of concurrent calls and so forth. I found teleyapper on

I hope this list is still active. I''m an experienced Linux Sysadmin, but I haven''t done much in the way of routing. Due to a decision made by my higherups, I need to jam a computer between my ISP and my LAN to do bandwidth throttling. My current setup: 1 Crappy Cable Modem (7Mb/768Kb connection) with a static IP. 4 servers (all have static, routable IPs) - One of which is

For an office that is using VoIP phones to connect to Asterisk, is gigabit ethernet really necessary for the Asterisk box to connect to the switch? I know that I won't even approach the limits of 100 Mbps, but would gigabit help with latency / collisions when several calls are underway? The fact is, anything going outside the office will be over a data T1, so intuition tells me that 100

Here is the situation: A T1 router going into an office which then plugs into the firewall box then into the switch. None of these devices support QOS.. Is there some sort of box/device that I can place between the T1 router and the firewall box which will allow me to prioritize voice traffic on this link? I can't change the T1 router to something that supports QOS because it has

We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. What we're trying to avoid is hardcoding the IP address in the ACL. We were trying to match by TOS set by Asterisk however it seems we've run into a snag where the packet TOS tends to get reset somewhere on our network. Has anyone had this issue? We're running Cisco everywhere inbetween (even the switches). Is

Has anyone tried out the Linksys RT31P2 with Asterisk? Seems like a really great solution for remote users... even supports QoS. Too bad it doesn't also have VPN functionality built in. Here's a link to the product: http://www.linksys.com/products/product.asp?prid=652&scid=29 -Ron -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list gurus, long story short we have firewall machine which is the default gateway for our clients and firewall send traffic out to Internet via cisco router. On cisco we have two serial interfaces 1Mb and 2Mb. On firewall #route add default gw xxx.xxx.xx.xxx (for 2mb) #route add default gw xxx.xxx.xx.xxx (for 1mb) and the same rule for Imb link route packets via these two links. However I

HI all, What is best way to be limited upload speed from LAN users. I read that it is possible to be done with IMQ interface or with limitation over gateway interface of router(eth0 in my "scheme"), but i cannot chose what is preferred way and need from advice. Please for advise, any example scripts or URL with tutorial are welcome :) I read couple times Linux Traffic Control.

I'm installing an office in a couple of weeks that will have some nice Polycom FX video units in the conference rooms. I'm thinking that with asterisk-oh323 http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/#section2 I should hopefully get the ability for phone users to dial an extension and participate in video conferences, or just simply phone conference with users in the

Is there any way to set Asterisk to choose what codec to allow for a new call based on current usage? In other words... be able to define a max number of ulaw calls, then after that only allowing g729? The idea here is that in general, a T-1 should be enough for our offices to have phone + citrix + some video (got good QoS in place already). But for usage spikes, user experience would be kept

I've got a SIP channel that appears to be hung up. It's an extension that records a .gsm file and fortunately the recording has stopped. I tried zap destroy channel but I guess that doesn't apply to SIP channels. Any ideas? I issued a restart when convenient but figure there must be a better way. TIA, -Ron -------------- next part -------------- An HTML attachment was scrubbed...

I'm running into a bit of a problem setting up conference calls. The box I rent at a colo doesn't seem to have USB hardware.... When I try to load usb-uhci I receive a "device does not exist" error. Which means I can't load ztdummy.... The system has a rtc clock module, so zaprtc won't work... (which I'm scared to unload rtc because I don't have physical access

For some reason a couple weeks ago users began experiencing garbled audio in one direction when dialing out via our VoIP provider. This happened at multiple sites simultaneously. The VoIP provider doesn't think it's their problem. If I switch to another codec so that Asterisk transcodes everything is fine. On conference calls (where Asterisk gets in the middle to relay ulaw to all

Hello. I'm try to set up asterisk for making outgoing calls with oh323 channel driver version 0.7.1 with Asterisk CVS-1-01/09/05-01:41:37. Our provider uses Mera MVTS softswitch and supports only H.323. We don't use gatekeeper for connection but provider requires SOURCE PHONE NUMBER for route out calls and I don't know how I can specify this number. Call with this string exten

It seems to never fail - after 3 to 5 minutes SIP -> IAX calls drop audio on one side. I place a call out through voipjet, and call quality is flawless. However a few minutes later the person who I'm talking to can no longer hear me. I can still hear them. What should I look for to resolve this? Has anyone else had this problem? Using last night's CVS this problem still exists.

Hello all. I''ve been struggling to QoS VoIP at our site and have a successful implementation at this point. Basically I had to set aside enough bandwidth for VoIP by placing all other traffic behind an HTB (multiple classes and queues behind it). Everything is fine. Here''s the diagram: ------- | eth | ------- | --------

Why not try something like Pebble linux from http://www.nycwireless.net/pebble It is a stripped down Debian install that is aimed at running a wireless hotspot but it is just Debian and you can install whatever you want. It fits on a 64MB flash card but if you install almost anything you will want a 128MB one. It does come with iptables and iproute2 tools as I recall. Patrick

Drive space for voice mail I've looked in the dimensioning information on voip-info.org but can't find any hard information on the amount of drive space the various codecs use. Since we would eventually like to support web-based voice mail retrieval, I'm thinking of the wav format. I've specced out 2x160GB drives in RAID-1 (software RAID via Linux) for the box. It will be