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...re difficult but if the design could be agreed upon at least those of us in between a rock and a hard place on this could decide to sponsor development, offer a bounty etc. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail

...<= seedNumber) && (valueDiff > sup)) { #error seeds[seedCount] <- fcsPar[k] seedCount <- seedCount + 1 } } sup <- sup / 2 } many thanks. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail

...lt;= seedNumber) && (valueDiff > sup)) { #error seeds[seedCount] <- fcsPar[k] seedCount <- seedCount + 1 } } sup <- sup / 2 } many thanks. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail

...about 5-10 seconds 5) incoming call drops This happens every time. Has anyone encountered the same problem? Would appreciate any suggestion. Asterisk version: 1.4; Polycom phone: IP301 /Why Tea Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail

I was wondering under what conditions Asterisk will hand off a call to another switch. I'm trying to verify that my local PSTN's Coppercom switch operates correctly... and wanted to know how to get a call REFER'd to another end-point. Thanks, -Philip

Hi, I have asterisk register two users (client-1, client-2) with a SIP proxy. I have the same two SIP client registered with asterisk. Now my dial plan setup is such that any call from client-1/client-2 is forwarded to the SIP proxy and the SIP proxy then takes the routing decision. Calls coming from SIP proxy will dial out the respective user. Asterisk is required to stay in the signaling as

...-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Murphy Software Developer Digium Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail

A call comes in from the PSTN, Asterisk answers it, it goes to the directory, and then to the extension the caller designates and the user at that extension answers. The user at the extension then wants to transfer the call to another extension; on the Cisco 7940 they push the ?more? soft key, then the ?Transfer? soft key, then enter the extension number they want to transfer to, and hit the

Hi List, I have a scenario that I want to try out (we potential have a client who would need this), but I am as of yet unable to find much help with it. What we want to do is have an asterisk box with a large number of extensions (1000+). This asterisk box will have approximately 3 SIP trunks setup back to providers. What we want to do is to be able to define groups of extensions that use