similar to: Router Recommendations Please

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 anyone

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

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

Grasping at straws here...is anyone using a Dell PowerConnect 2224 24-port unmanaged 10/100 switch in a deployment? I have two separate asterisk installations with bad one-way audio where the only common elements left are the Dell switches and Polycom IP-500 phones. Two different ITSPs - one location uses IAX2 to the outside world, the other SIP. Two different bandwidth providers, one SDSL 1.5

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:

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

All, I am trying to configure hdlc support on T100P Digium card - everything seems ok... but it just does not work at all. I was able to compile all drivers, the light on the card becomes green when I plug T1 link.... but I even can't ping default router IP... there is no data coming back to me, so I am confused completely now. Also I tried both protocols - raw hdlc and cisco hdlc. The

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

I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the

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

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

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 | ------- | --------

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

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.