[asterisk-users] TDM2400P Unable to set SW Companding on channel ..

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

[asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short

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 [9613070...@direct:1]

[asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short

Hi: Firewall is disabled ,so no need to worry about firewall,but i dont know where to check rtp settings and what do i need to search for ,can you guide me please. -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] TDM2400 card FXS problems

HI: I had this problem before with TDM2400P but with fxo modules and VPMADT032 (echo canceller),there was no audio at all.but then i unpulgged the ehco canceller module (VPMADT032) from the TDM2400P board and started the server  and then  i didnt face this issue any more. In your case  first

Re: [asterisk-users] R: rtp.c:883 ast_rtcp_read: RTCP Read too short

-users-boun...@lists.digium.com] Per conto di wassim darwich Inviato: giovedì 28 gennaio 2010 21:41 A: asterisk-users@lists.digium.com Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short   Hi: Firewall is disabled ,so no need to worry about firewall,but i dont know where

[asterisk-users] Codec coversion

Hi: Is there any software or hadware for codec conversion on asterisk ,any suggestion will be appreciated.   Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

[asterisk-users] codec conversion

Hi: Thanks for your reply,ill give you my situation, iam using my asterisk box as a switch ,so my client is sending me ulaw and my voip provider only accept g723 ,So what i have to do is to receive g711 codec and convert them to g723 at asterisk ,i tried this before but i saw the cpu usage is

[asterisk-users] Callcenter open source program

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. -- _ -- Bandwidth and Colocation Provided

[asterisk-users] dialplan

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 to specific  extension (contain ,not start or end with), as i know i should set Gotoif command but i dont know what to

Re: [asterisk-users] dialplan

. If a space is desired at the beginning of the data, then put two spaces there; the second will not be skipped. --  Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 28, 2010, at 5:49 AM, wassim darwich wrote: Hi guys: i need to set an extension in my dialplan