similar to: what is "tcp window size" ?

Hi all, For some reason VSAT or Satellite Internet services are not mentioned (or searchable) in this list's archives. I thought I'd let you know that I tested Asterisk using IAX (not IAX2) to make a phone call from an analog phone hooked up to an Asterisk system behind a Linksys router connected to a Gilat VSAT satmodem, and it worked. The "other end" (gateway) is a

Hello. I'm living in the country side where the communications are very very poor. My only choice is Satellite connection throw the French company Eurona which uses the network deployed by Skylogicnet. In general the latency is very bad (~800ms) and the network is very unstable. I have been investigating in the ISP routers which are in my path and there are many hops which are done in a

Hi All, Was troubleshooting some odd VM network issues and discovered that we're seeing dropped packets + retransmissions across multiple domU OS's and dom0 hardware platforms. xendev01 ~ # tshark -R "tcp.analysis.retransmission " -i vif7.0 Running as user "root" and group "root". This could be dangerous. Capturing on vif7.0 3.054257

I have a tc script which splits the bandwidth in 8 leaf classes based on IP filtering. The script looks like this: tc filter add dev $LAN_IFACE protocol ip parent 1:0 prio 1 u32 match ip dst 192.168.0.121 flowid 1:11 The separation works excellent for downloads, but for uploads, it is ignored... Can you tell me how to deal with it? Thanks in advance, Vlad Mihai

Hi all, Anyone has experience using * through 128 kbs (or bigger) satelite link? In particular I am interested to hear how many calls could be put through 128Kbs satelite link simultaneously? Ta SJ

Folks, We have a setup where each mobile node connects with 1 or more tinc instances (over different links) to a central node. tinc is running in switch mode. The link is chosen by setting the IP address on the active link's interface, and the central node sees this after the first packet on the link, and moves the MAC address to a different 'ethernet port' (link). This works really

Hi there, We use tincd over VSAT connections, and if the contract has been reduced to a bare minimum -and- the ship is moving, latency reaches 2 - 5 seconds (yes, seconds). Generally this is not a problem if you are patient (and most smart phones are), but tincd is slightly too fast in deciding that it is not going to work. We also use the tinc connection to check whether any traffic is possible.

Hi, I''ve a vsat internet link and I want to know if I can make VOIP calls under it. ( Ping in the best case reach 900/ 100 ms by sattelite effect ) Can with SIP protocol make voip calls ? Thanks andres

Hi, I have a PC that acts as a VoIP gateway using an internet telephony provider to make phone calls.. Data between me and the provider will always be from one IP address on my internal network to the provider.. I have a 1Mbps/256Kbps ADSL line and the voice stream is highly compressed requireing only 30Kbps of bandwidth.. The problem is that when my Mozilla Mail client polls for mail or

Hi All, I am investigating the deployment of VoIP/* in Eastern European areas where there is no PSTN infrastructure. As you can understand DSL/Cable connections are a dream. The only option is satellite. Does anyone know of any satellite providers that have low enough/acceptable delays for VoIP? Thanks, Yiannis.

Hello Everyone, I've successfully registered my g729a licenses. When i try to load the module from asterisk Cli i got the following error *Error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied* * loader.c:795 load_resource: Module 'codec_g729a.so' could not be

Hello... I'm involved in a project whereby a live audio input from the FM radio station is to be sent to an audio streaming server, which will stream the audio to the remote sites thru the VSAT (satellite based) network. The remote sites are to rebroadcast the transmissions (FM station rebroadcast), as most of the expected listeners do not have access to the internet. Is icecast2 capable of

Hi All; Can I run the IAX on TCP port instead of UDP port? If I ran IAX in TCP port, and in case my network was having a lot of users doing browse on the internet and downloading, so in that case and if the IAX used TCP port, so the voice will be better than using UDP (because in TCP the lost packets will be resend while in TCP it will not which will cause the voice to be cutting)? Same thing

Hi all, ------ I tried to use * over satellite, but all my effort did not succeed. The Asterisk is behind the VSAT and is resposibel for alle the SIP clients in a field location. The clients are notebooks and PDA's running SJPhoen for Windows and PocketPC. Unfortunately I could not find any Linux Client wich worked satisfying. SJ LAbs promised a Linux Version at the end of August but they

Hello Guys, In the case of a multiserver environment for outbound automatic calls, can you share you experience and preference between call files and ami originate ? thanks -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -------------- next part -------------- An HTML attachment was scrubbed... URL:

I found following paragraph in the man page of iproute2. equalize allow packet by packet randomization on multipath routes. Without this modifier, the route will be frozen to one selected nexthop, so that load splitting will only occur on per-flow base. equalize only works if the kernel is patched. ^^^^^^^^^^^^^^^^^ Now, where

Hi Everyone, Can you tell me how many concurrent TDM (Dahdi) calls that a single asterisk box can handle. Configuration is as follow : Quad core Xeon 3 GHZ, 4Gb RAM, asterisk 1.6.2.9 Also do you know a good tool to stress out asterisk? Kind regards -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* --------------

Hello, I have 2 sites. One(Site 1) has an asterisk PBx and the Other(site 2) has 2 remote soft phones. The latency btw both sites is btw 500ms-700ms. I know this is a shot in the dark...but are there ways of improving the voice quality for the remote calls(btw site 1 and site 2), Other than increasing bandwidth? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994

I want to thank the list and Andy Furniss for helping me get started with TC. Here is my current script with some additional comments. Please note that I can''t really endorse anything below, as I''m new at this. I can tell you that using the monitor_tc_top.pl script, I can see traffic going into all these classes and that my users seem very happy with the results (we are 57

Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Thanks a