similar to: Re: update - 512 Simultaneous Callswith DigitalRecording

The simplest solution and the one already implemented in linux, tmpfs. It would be best to allocate 4-8GB to tmpfs on /tmp and let the kernel do the work it was designed to do. And you would not be limited to PCI bus speeds. The DDR2800 is about 12GB/sec. Some would say "overheads, etc, etc". Agreed, even at 95% loss (doubtful) you still get higher badwitch then PCI bus/hard rive could

I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? TDM800P and HPEC The TDM800P is an 8-port analog telephony interface card, so it fills the gap between Digium's 4-port and 24-port cards. Analog phones and POTS lines are going to be with us for some time, and demand for support for them remains high. The

Hi, How would one activate/deactivate hardware echo cancellation on the TE406 card? Can it be done per channel? I'm going to run TE406 in the following scenario: ISDN -> TE406 -> PABX I understand from Steve Underwood's site that echo cancellation is not good for faxes (and they do that themselves). So what I want to do and bypass echo cancellation for selected extensions before

Hi, Anyone using SpanDSP with Digium TDM o TE cards to receive and email Faxes? Thanks, Javier Ergas R. Director General de Tecnolog?a Pibix Telefon?a IP http://www.pibix.cl -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060615/ed77e5d1/attachment.htm

Following is the last few lines of output when i try to launch Asterisk:- [app_zapscan.so] => (Scan Zap channels application) == Registered application 'ZapScan' [app_saycountpl.so] => (Say polish counting words) == Registered application 'SayCountPL' [func_cut.so] => (Cut out information from a string) == Registered custom function CUT == Registered custom

Can anyone shed some light on this, I have two asterisk boxes using heartbeat for failover. Sip traffic works just fine with the virtual IP but IAX does not. For example on my servers one server has the following: eth0 = 192.168.1.95 eth0:0 = 192.168.1.2 the other server has: eth0 = 192.168.1.220 if the first "Master" server goes down the second server will take that virtal IP for

Hi, I just read a pressrelease from VON that Digium will soon be releaseing a couple of new cards. What got me interested was: "The TE420P and TE415P support 128ms of G.168 (2002)-compliant echo cancellation across their entire 128 channels." Does anyone know when thease will be released and what they will cost when released? Thanks!

Hi Can TDM400P detect polarity reversal on FXO module? We have C.O. lines that reverse polarity on Answer and release. Thank You This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you

HI, Does anyone know if there is a PCI-X 4 port PRI cards available on the market? If so, have anyone used it and how reliable they were? Any help is appreciated...

Hi, I've installed an TE406p, asterisk1.2 on tyan opteron board. After installation there is no interrupts from TE406p. Is this board stable? Should i change * version to 1.0.9? Regards, Jason __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Hi, I got a problem with PRI that I'm not sure how to solve. Asterisk sits between PABX and PRI. PRI is span 1 and PABX is span 2. After every single call (no matter in what direction) I get "pri_fixup_principle: Call specified, but not found?" and "pri_dchannel: Hangup on bad channel" messages and the channel in question is restarted. As far as I can see, all

Its slow :) It will give you some delays but it will not be noticeable (most voice files are 5-100kb, so it should be ok... But writing to them.. Not sure.. It should be ok as well I'm guessing as kernel will provide some caching (since you have G and not GS it has less ram, so maybe chaching is not an option) Best would be to get WL-500G. It has a USB port so you can plug a USB memory stick

I had CC readers going over the internet (with pings over 80ms) connected to Linksys PAP2. It was only successful once every 3 attempts. I had 100% reliability when it was connected on LAN. Timing is an issue, if you doing everything on LAN it should not be a problem. Just make sure you use G.711 protocol. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com

I'm attempting to setup Asterisk 1.4.0 CDRs to use MySQL. Modules show like cdr_mysql.so tells me it is loaded. Reload cdr with MySQL started or stopped makes no difference in the errors. Ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070220/55700f37/attachment.htm

Hi All, In previous mail lists, people talked about a solution to record large amount of simultaneous calls. And then it seems that RAM disk solution was the best choice due to the I/O bottleneck of Hard disk (System). Please find the previous discussion as follows: http://lists.digium.com/pipermail/asterisk-users/2005-October/120930.htm l

Guys. I have an issue with a te110p card and also some tdm04b cards on the same system: Zttest returns this for the tdm04b cards: [root@mollendo ~]# /usr/src/zaptel-1.2.4/zttest 38 -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.000000% 8192 samples in 8192 sample intervals 100.000000% 8192 samples

List users, Over the last few days we have been working with MCI's development lab to test our Asterisk setup. We were using a piece of hardware called an Abacus 5000 that is capable of creating and terminating thousands of SIP calls. Initially, we could not get past 64 simultaneous digitally recorded calls without having call quality issues including dropped calls. We identified an

Hi, I thought I'd check in with people here. Lets say I wanted to put together a Hylafax server, and an Asterisk server on the same box. And, for the sake of making things simple, I wanted to use a single T1 board for both.... Is this possible? I see Eicon boards are supported by both software, so could I have Hylafax and Asterisk installed on a server with an Eicon card, with channels

Andrew Kohlsmith wrote: >I wasn't talking about bandwidth but rather lengthy >Dial() commands... > >exten => s,1,Dial(SIP/someuser&SIP/someuser&SIP ...... > >kind of thing... seems awfully unwieldy. That's why you would stick the members into a global variable [globals] DIYCALLGROUP => SIP/111&SIP/112&SIP113 etc. then dial using

If you have a rjs template with 10 lines of code. Is it possible to make then run in order, one after the other. Is this possible with rjs or are you stuck with the simultaneous execution of the javascript. I don''t want to resort to the delay method. Charlie bowman recentrambles.com -------------- next part -------------- An HTML attachment was scrubbed... URL: