similar to: Asterisk crashes my router!?

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

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 have been trying to optimise my ADSL connections for VOIP. Funny things were happening - for example increasing the ping packet size by 50% had no effect, but then adding one byte had a major effect. It took me a while to figure out that I was seeing the effects of the fixed ATM cell size. This is probably obvious to some of you. For the rest: ADSL uses ATM as its transport. An ATM

Did anyone see the article? It''s the first time I really noticed that these little Linksys routers are such a fully fledged linux machine with a decent processor and a replacable firmware. I am now itching to get one to replace the multipurpose firewall desktop machine. Has anyone experimented with the current state of the firmware and how advanced you can get with tc rules? For

I'm in conversation with Draytek's pre-sales dept.............. Here's the most recent reply: <Hello, We really don't know of anyone who has run an Asterisk server on a Vigor2900. There are doubtless people around, but it's relatively rare. Most people don't run SIP servers. Regards,> All I want to know is, if I buy one of these routers, will it break my setup or

I have been trying to read everything I can find on Sipura 2000 and Asterisk. I am trying to make the Sipura-2000 act as two analog lines off my asterisk. I have followed (what I believe) the example on http://www.voxilla.com/Article39.phtml and I still can't get my Sipura to register with my Asterisk server. I can re-config my Sipura to talk to fwd, or voice-pulse connect and it works

Hi I''m using wondershaper like script. But noticed imperfect scheduling. By googling, I found some patches that takes the aal5 atm overhead in the scheduling algorithms. http://www.adsl-optimizer.dk/ADSL-optimizer/ Before trying it, has anyone tried these patches? they apply on 2.4.29 kernel and 2.6.9 TC. I use centos4 with a 2.6 kernel, so I can''t try these easily. Thanks

Hi, Is there some script which can be called from a * extension to playback the recent incoming callers on a particular PSTN line? In the UK 1471 is a BT number which plays back the most recent callers number, it also gives you the option to call this number back (now charging you for this service too!). Is there anything similar in asterisk-land? thanks Mike

Hi, I'm looking for opinions on the "best value" router to use for home offices. It should work for a scenario in which there are 3 computers and 2 SIP phones, handling QoS so that the phones always have higher priority traffic than the PCs. (and not rely on the phones to do the QoS because some PCs may not be connected to the phones). QoS could be based on destination and source

Hey, I''ve been scouring the web these past few days trying to find a suitable technique to load balance my network. Unfortunately I think my setup is very uncommon, so I''m looking to you guys to see if you can help. I currently have 2 ADSL modems, provided by the same ISP and terminating on the same gateway (I think having the same gateway is what is causing the problems, since

Guys Considering the festive season is upon us, thanks to everyone contributing to the list and helping all the readers with your great input! I don''t want to mention names, I''ll most certainly leave someone out. With this mail I''d like to test some theory on bandwidth management, with my own successes and failures during the past year. Sharing a link between 200

Dear list, Shorewall is running here with 2 ISP''s: ISP1: corporate ADSL-line with fixed set of IP''s ISP2: fast consumer-grade cable-connection with higher bandwidth All our main traffic (web, e-mail) is routed trough ISP1. Only for special purposes (frequent large ftp-transfers) ISP2 is used, configured trough tcrules. ISP2 is not so reliable as ISP1 (duh) and they sometimes

Hi. I wrote in a reply to a mail on here recently that you can''t set mpu (minimum packet unit) on HTB as you can on CBQ. I''ve just noticed that there is a patch on devik''s site which does mpu and overhead. http://luxik.cdi.cz/~devik/qos/htb/ For dsl users mpu is, for practical purposes going to be 106 - overhead is still variable though, depending on packet size.

Hello all, We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond....) We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line

Hi Maarten, Yes, I'm behind ADSL with PPPoE but I don't think it is related with my case. When my client disconnects from IceCast server for 15-20 seconds, my active downloads are not interrupted. It currently effects only the route to my IceCast server, other routes looks uneffected. I'm pinging IceCast server continuously and checking trace route periodically. It is comething like:

Hi, I've got a client with two ADSL connections for redundancy. Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections? Or to use one connection as the main one, and automatically fail over if the first connection drops? Or does this kind of thing need a serious network switch? Thanks Dan --------------

Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is

I received this before, and it is because you are using the wrong context in the iax.conf. For example: The context must match the username in the register statement. Iax.conf... register => username:secret@iax.host.net [username] type=friend context=iax-in user=username secret=secret auth=plaintext host=iax.hust.net ----------------------------------------------------------------

Our users are experiencing some unacceptable delay when trying to have a conversation. The delay is so noticeable that they keep stepping on each others words and resort to calling the customers via cell phone. Here is the setup SDSL Connection (PPPoA) Speedtouch 610 SDSL Modem 3Com 2224PWR Plus Switch (phones on separate VLAN) 8 Cisco 796 Phones All connecting to a remote Asterisk Server. We

Hi, I've had an inquiry for a small UK call centre, mostly outbound calls. I get the impression they are mainly calling 3G mobile phones, monthly phone bill, with calls is approx ?5,000 for several feature lines. How feasible is something like this with asterisk? I guess one big question is which type of circuit to use, ADSL in the UK is only 256kbs upstream, some providers do bonding but