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Will M$ ever stop!!.. Whats the bet their telecoms products will use non-standard protocols.. I really wouldn't like to run a telecom system on Windoze in the first place.. Full Story.. http://news.zdnet.co.uk/communications/0,39020336,39117099,00.htm

Hi, i am recording a message, but it is always cut off at 40 secs. There are no time out configured. Gabriel -- The educated person is not the person who can answer the questions but the person who can question the answer.

I have set up a couple of test Asterisk servers and have never had a problem with sound. I've just done a fresh install on a dual 1GHZ PIII Asus box running Fedora Core3 with the Digium PCI Dev kit and following all the various Core 3 How-To's. I can make calls ok but when any sound is sent from the Asterisk box such as voice prompts and music on hold the sound is completely chopped up in

<P>hi everyone,</P> <P>I know this issue has been raised many times before, i think still the problem remains. When a call is made through a Zap channel, whether it is actually made or not (irrespective of whether, engaged, busy, or actually answered), asterisk logs it in CDRs as a call made. This makes it impossible to do an accurate billing. Has anybody found a way to overcome

Forwarded back to the list so others might get the benefit of the answers, and I get fact checked by others. -------- Forwarded Message -------- > From: Lee <leeb00@gmail.com> > Reply-To: Lee <leeb00@gmail.com> > To: Steven Critchfield <critch@basesys.com> > Subject: Re: [Asterisk-Users] udev or not? > Date: Fri, 10 Dec 2004 13:00:29 -0800 > On Fri, 10 Dec 2004

Supposed scenario: one PC(2GHz CPU), one card 4E1, and one Internet link. There is somebody he know (has experienced) how many concurrent call (Classical Phone->Voip) can handle Asterisk ? Thanks ! -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Can T400P be configured for nfas (one d-channel providing signaling for more than one span)? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030522/f2b637a9/attachment.htm

hi guys, do you have any suggestions on how to connect 2 TE410P via E1? (for simulation and testing purposes) asterisk1 --> TE410P ----> ? ---------> ? ---->TE410P -->asterisk2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030907/698cd499/attachment.htm

We have 1000's of Remote Call Forward #'s across the USA / Canada, which forward into 1000's of 800 #'s in our call center. Is it possible to automate a solution where Asterisk could dial a given number, record the first 3 seconds of the call, save it to disk, and then go on to the next number, and just do this all day long ? We need to regularly check that the numbers work, for

Why does this person have my e-mail address ? ----- Original Message ----- From: <markogift@astriholics.org> To: <hackerwacker@cybermesa.com> Sent: Tuesday, November 23, 2004 1:13 PM Subject: Gift for Mark Spencer > Hello everyone! > > We have been thinking about something that we could do for Mark > Spencer as a holiday gift. We have decided to try to orgranize a

Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a "basic" Linux/Asterisk system? Just re-boot and config. -- James Taylor jltaylor@metrotel.net 903-793-1953 --

With the increased traffic as of late, I'm wondering if it is time to split the list again. Specifically I am wondering if it should be split along the various VoIP protocols and zap hardware, then leave a general list that does configuration other than VoIP related? The hope is that those asking SIP or H323 questions could get help from the various supporters while the main list can deal

Greetings, I am running Asterisk with a T100P and a Zhone channel bank for over a month now. For the most part it works fine but from time to time (about once a week) the system will not let go of a line and will play the greeting over and over. Anyone calling gets a busy signal. If I reset Asterisk everything works fine. Has anyone seen this problem before and fixed it? If so what did you do?

from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx "restart now" >/dev/null 2>&1 or 10 7 * * * root /usr/sbin/asterisk -r -x "restart gracefully" >/dev/null 2>&1

I am trying to compile spandsp on my asterisk server, and it keeps failing out with the following t4.c:38:21: tiffiop.h: No such file or directory In file included from t4.c:41: spandsp/t4.h:62: error: syntax error before "TIFF" spandsp/t4.h:62: warning: no semicolon at end of struct or union spandsp/t4.h:63: warning: data definition has no type or storage class spandsp/t4.h:64: error:

I just updated to the new CVS and now I am getting the following error from chan_zap (modprobe wcfxo works fine): WARNING[16384]: File chan_zap.c, Line 577 (zt_open): Unable to specify channel 1: Device or resource busy ERROR[16384]: File chan_zap.c, Line 4781 (mkintf): Unable to open channel 1: Device or resource busy here = 0, tmp->channel = 0, channel = 1 ERROR[16384]: File

Hello, It is my understanding that on the softphone side, asterisk is only responsible for establishing the session between two phones. If this is the case, does it matter what type of audio codecs the two phones are using? And if it does matter, are there any codecs that cause problems with asterisk bridging two SIP connections? Thanks for your helpful input, Daniel

Hi, I've been browsing for FXO devices, and I'm really surprised at their costs. Why such devices are so expensive and somehow hard to get ? Samy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030805/a92ee327/attachment.htm

I'm going to install asterisk with four digium cards. Can anyone mention a brand that carries boards with 4 compatible pci slots? Thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200