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HI all: Iam planning to use my asterisk box as callcenter?,any one can advice me with the best callcenter open source program based on asterisk . ? Any help will be apreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100307/116f1b75/attachment.htm

Hi guys: i need to set an extension in my dialplan?in which?it divert calls if?the extension?contain specific series ,For example?: I need to divert calls which contain 3333 to specific? extension?(contain ,not start or end with), as i know i should set Gotoif command but i?dont know what to set after that,Any help will be appreciated.??? -------------- next part -------------- An HTML

Hi: Is there any software or hadware for codec conversion on asterisk ,any suggestion will be appreciated. ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100202/ee00a520/attachment.htm

Hi: I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys?, This is wht i see on asterisk console?: ? -- Executing [9613070741 at direct:1] Set("SIP/03070741-088bd470", "CALLERID(number)=96170707070") in new stack ??? --

Hi: I?Bought TDM2400P ,with 24 FXO ports , I installed? the asterisk 1.4.28 and?dahdi 2.2.0?and then i compiled them and configured all, i pluged into 8 pstn line into the Rj connector and then i got messages on asterisk console that alarm cleared on channels 1->8??, at this step everything should go fine ,but when i try to make a call?through the TDM?there is silence and no audio,and then i