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...et you set the interrupts yourself and you end up with interrupts being shared. =============================================================== Message: 26 Date: Mon, 25 Jul 2005 08:31:47 -0500 From: Eric Wieling aka ManxPower <eric@fnords.org> Subject: Re: [Asterisk-Users] Need Advice To: tarsi@binhost.com, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <42E4E9C3.2080806@fnords.org> Content-Type: text/plain; charset=us-ascii; format=flowed Nathan Pralle wrote: > However, for FXO ports, I'm using the Digium Wildcard...

Is there a guide anywhere which runs through how to set up asterisk with mysql? I've looked and almost all the document misses out relevant information. Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050920/1ce45adb/attachment.htm

HooDaHek 0.6 has been released. So soon, you say? Well, the best laid plans of mice and men... Steven BerkHolz is a pretty sharp stick and said to me, "Why don't you have HooDaHek change the CallerID when it looks up the name in the database on an incoming call?" Much head smacketh ensued, and as I made that change for Steven, I noticed that I had the way wrong version of

I see that the regular site for the Asterisk Perl AGI module is down (or something), so I've made an alternate download from my site: http://www.nathanpralle.com/downloads/asterisk-perl-0.08.tar.gz (just in case somebody was looking for it) Nathan -- ------------------------------------- Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com

Hi all. Searched the archives but couldn't find anything on this: I want to track 2nd incoming calls on a single line but don't want to pass the Call Waiting pips along to the engaged user. IE: I want Asterisk to detect that CW is currently being transmitted on the line, and track it, but not pass it on. Is there a way to do this? TIA, Nathan -- ----------------------------

I'm using get_data in an AGI script and am having a problem when, after a long time in my IVR, when I ask for a 10-digit phone number, the first few tries are always invalid -- the number it reads back is very strange, almost like the DTMF tones from other answers were being cached and then dumped on the call to get_data. Anyone ever experienced this before? I have to do some major

Hey folks, Anyone know of companies selling bulk SIP adaptors (phones, adaptors, etc.) or has the list ever considered doing something like a bulk buy? I was just curious...I'm looking to get another 5-6 Grandstreams or similar and I figured I'd ask the list. If we found something that lots of people wanted, it probably couldn't hurt to contact a company and ask for bulk deals.

Hi all. Done some research, Googled a lot, but can't find out if there is a USB FXO adapter that works well with Asterisk. If someone knows of one or has used one, I'd be very interested to hear about it. Many thanks, Nathan -- -------------------- www.nathanpralle.com --------------------