similar to: minimum hardware

Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

Hello. I had grandstream working fine to FWD through my firewall. Now I want it to talk to the asterisk server. Did lots of reading, attempts but I keep getting registration errors even though I can call to/from the sip phone from an analog phone on a tdm400 card. Basically. grandstream = 192.168.1.70 asterisk = 192.168.1.1 The error I see is ;- -- Executing Dial("Zap/2-1",

Hello. Is this normal. Or does it mean there is a problem ? ------------------------- stop now Beginning asterisk shutdown.... Executing last minute cleanups == Destroying any remaining musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Yuck! Error in buffer handling...: Broken pipe Yuck! Error in buffer handling...: Broken pipe Asterisk cleanly ending (0).

I have a Digium TDM400P card with 1 FXS and 1 FXO module running on my FreeBSD 6.0 server. While I am on the road, I would like to save on costs by using a soft-phone from my laptop to call in to a telephone connected to this card. I installed both Asterisk and Zaptel drivers from the ports, but still haven't done anything with the configuration files. What else do I require, and what is

Hello All, I have a very small setup of 4 users and a X100P. Asterisk is currently running on an Athlon 1800 but the server it is running on is also our imap/web/mail/development/samba server and we are having a few issues with asterisk which I believe is down to to many tasks. What I intend to do is build a box just for asterisk(Ill call the old one obelix). I have a few machines in the loft

Hi. I have been searching through old threads and various forums, but can't seem to find out what the consensus view is here on the best type of FXO hardware to use for a home asterisk implementation. Specifically, should I buy a FXO ATA like the Sipura 3000, or a PCI form factor port like the Digium TDM400 series. I have read horrible things about problems with the X100P, so I have ruled

I read a dataset with times in them, e.g., "09:31:29.18761". I then parse them: > all$X.Time <- strptime(all$X.Time, format = "%H:%M:%OS6"); and get a vector of NAs (how do I check that except for a visual inspection?) then I do > options("digits.secs"=6); > all$X.Time <- strptime(all$X.Time, format = "%H:%M:%OS"); and it, apparently, works:

>Kenneth Arnold wrote: >For all that verbosity, my proposition is a simple >question: what is the mimimum set of changes needed to make a killer >editing format out of Ogg, and are those modifications small enough >that streaming-Ogg and editing-Ogg can be unified such that one is >just a special case of another? I think this is an important concern >that is best addressed

Hi all !!! I would like to install asterisk in Xen domU using TDM400 hardware. Somebody know a howto or tutorial about that ? Thanks in advance roberto -- Ing. Roberto Pereyra ContenidosOnline http://www.contenidosonline.com.ar

Hello. Well I finally rx'd my dev kit (new batch of TDM's apparently. I'm on Mandrake 9.1 There were no hardware install instructions, it would have been nice to know whether the 4-way power connector was to be used or was for some other future or expansion purpose. It came with a floppy disc, no label and it wasn't even write protected. The only readme file was

Dear R-er, For a non-linear optimisation, I used optim() with BFGS method but it stopped regularly before to reach a true mimimum. It was not a problem with limit of iterations, just a local minimum. I was able sometimes to reach better minimum using several rounds of optim(). Then I moved to optimx() to do the different optim rounds automatically using "Nelder-Mead" and

Address lookup canonical name asterisk.org. aliases addresses 216.27.40.102 Service scan FTP - 21 Error: TimedOut SMTP - 25 Error: ConnectionRefused HTTP - 80 Error: ConnectionRefused POP3 - 110 Error: TimedOut NNTP - 119 Error: TimedOut digium.com is ok though Address lookup canonical name digium.com. aliases addresses 216.207.245.1 Service scan FTP - 21 Error: TimedOut SMTP - 25

Hi, I tried to run the make progdocs, but the first time, it said, I have no doxygen installed. So I compiled the latest release and reconfigure the asterisk. And after it, ut sucessfully started to build the docs. But it took a lot of time, So finally I aborted the process... Is there a way to disable doc creating? The --disable-xmldoc is enough? Thanks! On Dec 2, 2016 3:36 PM, "Joshua

Anyone tried to build * + h323 to rhel3... I have to problems in the process... a) Zaptel would not build - a whole bunch of errors about kernel... b) make progdocs failed with reference to dot - check your installation. Do I need the zaptel ?? I will not be using any interface cards.. I'd like to make progdocs - any suggestions there? Latest cvs on everything.. Thanks

Hello. I can't find a gsm plugin for XMMS. How do Unix, Linux, BSD users listen to gsm samples ? Regards...Martin -- While you don't greatly need the outside world, it's still very reassuring to know that it's still there.

Hello all, After installing Asterisk, i have installed the docs by "make progdocs". But i don't know where to locate this documentation. please Help. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070810/ceb95948/attachment.htm

Hi, I've updated my Asterisk 3 times with : cvs checkout -r v1-0 zaptel asterisk asterisk-addons and then do cd asterisk make clean && make && make install make samples make progdocs and then when I run Asterisk I get : Asterisk CVS-v1-0-02/11/05-01:46:25, Copyright (C) 1999-2004 Digium. Is this a bug in CVS handling or am I doing something wrong ? How to check which

Hi all, We have tested both tdm400 and BRI 1 port HFC based with mISDN driver and with both cards we have the same problem... so probably it's due to our misconfiguration, but we couldn't find anything useful on google. With TDM400 we can place and receive calls, but if we receive a call from TDM400 and we reject it from a sip phone , asterisk send the hangup but the line continue to

Hello. I'm seeing this at the console. NOTICE[81926]: File chan_sip.c, Line 5119 (handle_request): Registration from '<sip:marrandy@192.168.1.1>' failed for '192.168.1.70' What's this all about ? Regards...Martin -- Osborn's Law: Variables won't; constants aren't.

Hi, all Did anyone tried TDM400 card on old main board (Intel 440LX chipset -- PII)? The reason I am asking is because TDM400 needs PCI2.2 and main board is PCI2.1 I do not want to upgrade yet. Thanks, Rudolf