search for: com_wrapper

http://dev.mysql.com/doc/refman/5.0/en/too-many-connections.html Max connections is set to 100+1 by default. Maybe you just need to raise that number. Otherwise, figure out why connections are staying open if they are. Thanks, Steve Totaro http://www.asteriskhelpdesk.com/component/option,com_wrapper/Itemid,37/ KB3OPB > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Doug Lytle > Sent: Wednesday, May 02, 2007 7:20 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Sub...

I'm looking to hear from any application developers in Argentina specifically or other South/Central American countries. Please understand this isn't a dial plan or remote installation I'm looking for but an actual application developer. If this fits your description please email me details on; Size of company (number of full time/versus contract staff) Location Previous

...gives them an IP, my script is run automatically to change the firewall rules and start recording their presents, or if it''s an unknown user it will add their MAC to a list of unknown users. More details of the application can be seen here: http://www.tcn.bowenvale.co.nz/component/option,com_wrapper/Itemid,23/ Cheers Don -- Don Gould www.thinkdesignprint.co.nz - www.tcn.bowenvale.co.nz - www.bowenvale.co.nz - SkypeMe: ThinkDesignPrint Upcoming events: Software Freedom Day - 16 September 2006 - www.softwarefreedomday.org _______________________________________________ LARTC mailing lis...

hi List, i've been trying to get festival work on my 1.4.4 *box for the last 3days, i've used the tutorial on this page http://www.voip-info.org/wiki-Asterisk+Festival+installation with exactly the same line in my dialplan just to make a test now when i try to call( dial 555 ) from my softphone i get this message on festival server debugger: server Tue May 8 11:36:53 2007 : Festival

Sorry for disturbing you, but we have some problems with an installation with multiple (84) T1s from Quest. Now, our Problem is disconnected numbers are reported by sending in- band channel alert message and the B-Channel will have the tri tone and respective message but the line is never "picked up" and stays in ringing when dialling. So disconnected numbers are never detected as

Hello all, I have a set up that answer my customer. and its working well, however, the number of call to technical dept is what i want to reduce. I want all call to get to voice prompt except that that enter when minutes is 15, 30, 45, 60(in multiples of 15 minutes). how can i achieve this and what application can i use to get this done. I will be glad, if someone can give me a hint on this.

So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry

Hi, We have a PRI connection & when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 & RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats