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Hey there. I've been working on a project to integrate Asterisk with Exchange Unified Messaging via sipX using large parts borrowed from: http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html ... and everything works surprisingly well. The one problem I have is MWI, or a lack thereof. Exchange 2007 doesn't support MWI of any kind (!), so I've been looking into

...ly be a lot more 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

...if((seedCount <= 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

...f((seedCount <= 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

Can Theora be streamed IP Multicasting? Can Cortado support IP multicast? if not, could some add IP multicasting to Cortado? Find a better answer, faster with the new Yahoo!7 Search. www.yahoo7.com.au/search

...on of area. Specifically the data i'm trying to analyse relates to how much fuel was burnt in response to fire and recorded as a percentage consumed. Any ideas or suggestions would be greatly appreciated! Ta heaps Jen Get the name you always wanted with the new y7mail email address. www.yahoo7.com.au/y7mail [[alternative HTML version deleted]]

...d B-party for 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

Hello, I'm having troubles while using the "Goto" function in a realtime extension. Here is the error message : -- Executing Goto("SIP/siemens1-081f56b0", "script_13_0|s|1") -- Goto (script_13_0,s,1) [Feb 7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel 'SIP/siemens1-081f56b0' sent into invalid extension 's' in context

...w bug reports with only one response, and that was a long >time ago. Certainly looks like their focus is elsewhere, which is a real >shame because as far as I'm aware there is nothing else like it. >-Phil Get the name you always wanted with the new y7mail email address. www.yahoo7.com.au/y7mail

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

Has anyone tried to used VB6 to communicate with the Asterisk Manager? If so, would you be willing to share some basic code showing your approach to getting connected and parsing results? I've got a Telnet control that is allowing me to connect, authenticate and see the "flow" of status, etc., but I'm sure there is a better way to do this without using Telnet (maybe not?). Any

...pi-digital.com-- > > 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

I am having issues with transfers (SIP/REFER) using Asterisk 1.6. You will find the SIP debug below. There are three phones in this setup. 5253 and 5258 are Aastra 53i telephones, 101 is a standard phone connected through an Audiocodes gateway. All phones are registered in context "phones" and are set to not allow reinvites. All phones can dial each other directly. The dialplan