similar to: FXS on TDM12B suddenly stopped working Properly

Hi: I was running TDM12B. Both FXS and FXO were working fine. Then all of the sudden FXS had problems. When I pick-up the phone and dial any number, FXS doesn't respond. I just keep hearing the normal signaling line tone comming from the FXS. I changed the FXS module and it had the same problem. I changed the the TDM card and installed different FXS and nothing changed. I appreciate any

Can someone please help me with my config settings for my TDM12B. I have got the FXS port to work fine, but I cannot get asterisk to startup with any settings for the fxo ports. It appears to be something related to the message "Signalling requested is FXS Kewlstart but line is in FXO Kewlstart signalling". However, I have tried what appears to be every possible configuration option. I

Hi: Please advice me of a voip provider with reasonable reseller program. I was using voipjet and it has a lot of problems. Did anyone experienced asteriskout.com service? They have good prices. Regards; Chawki Hammoud ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

Hi: I had an fxo card from Digitnetworks and it was working fine on my Asterisk box. I then replaced it with TDM04B. I changed the zaptel and zapata to include the four channels. When I run ztcfg, I get configuration errors: Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04:

Is there a way to hide the callerid on analog line on outgoing calls. Any ideas whether it could be done through configuration, a script or hardware. Thanks; ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

Hi: I have one TDM40B and one TDM04B on my Asterisk box. Both were working fine. Then, all of the FXS ports started to make echo sounds when I make FXS to IAX or SIP connection. All of the FXS ports fail to make bridging to the FXO channels. And when I try to make a call from the console to the FXO ports, I only hear static noise. Any suggestions about the source of the sudden change in

Hi: Astcc is working fine, except for one thing. It doesn't give the called party enough time to answer the phone. If nobody picks up in two rings, astcc reports back no answer and hangs-up. The only instant NOANSWER "value" was mentioned in astcc.agi script is: elsif ($res eq "NOANSWER") { $res = &mystreamfile("astcc-noanswer");

Hi: I want to connect a wi-fi phone to my Asterisk box through a wi-fi AP so I can make voip calls. please send me your recomendation about what wi-fi phone I should be looking for. Anybody tried the HOP1502 Wi-Fi IP phone. Its listed price $39. Regards; Chawki ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football

I appreciate any recomendation of a simple CallingCard Application and resources of users manual. __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/

Hi: It's the first time I have this problem. No matter how I configure my two IAX machines the connection is rejected. "chan_iax2.c:5550 socket_read: Call rejected by ****: No authority found" iax server A: [saad_out] type=peer host=hostip username=username secret=secret disallow=all allow=gsm iax server B: [guest] type=user username=username secret=secret context=tele

This is my second attempt trying to get help and I am hoping someone can. When the musiconhold extension is matched, Asterisk attempts to execute musiconhold and stops right away, this is what I gets: Executing MusicOnHold("OSS/dsp", "") in new stack -- Started music on hold, class 'default', on OSS/dsp -- Stopped music on hold on OSS/dsp Is there a file that

Hi, I have a TDM21B (A new TDM20B that just arrived plus a X400M that I already had). This card sometimes works, and sometimes only the FXO module is seen. By "works", I mean all 3 modules have the green lights on the ports on and when zaptel is loaded the log shows: Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module

I have a TDM21B (A new TDM20B that just arrived plus a X100M that I already had). This card sometimes works, and sometimes only the FXO module is seen. (correction to previous message that incorrectly listed X400M/S400M when I'm using only the single port versions of these modules (X100M/S110M). By "works", I mean all 3 modules have the green lights on the ports on and when

Hi all, I have hit a wall configuring a TDM400, I have set these up before without issue but today I just can't seem to figure out what I am doing wrong. On an incoming call the following is produced in the Asterisk console with verbose 4 -- Starting simple switch on 'Zap/2-1' Mar 22 16:12:34 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... Mar 22

Hi Mary Christmas to y'all ! I am trying to configure * with one TDM and one ISDN and Bri-stuff, all is ok with only TDM but when add ISDN I get following error: I have tried to switch fxs/fxo but then I get error also without ISDN, card is configured from Digium with FXS(green) modules are closest to the bracket. ZT_CHANCONFIG failed on channel 1: Invalid argument (22) Did you forget

Hi: The phone company here use Europe telephone system. They offere ISDN lines with multiple channels, starting with two and they add channels on demand. Am I write to understand ISDN of more like E1 or T1 line. If not, what's the difference. __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone.

I apparently have a dead FXS module. Is there any kind of test I can run on it (on a live system) to determine if its good or bad? The telephone has no dialtone, gets no calls. It has been working for several months, and just quit yesterday. Thanks! Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO

Hi All, Anyone have any idea what could cause incoming calls on a SIP channel to no longer be able to use DTMF? DTMF on incoming calls on zaptel and on local SIP softphones and ATAs all work fine. Nothing gets registered in the CDR or on the console in verbose level 10, it just times out. I haven't changed anything on my part and can't get through to Viatalk tech support to ask them

On 15-06-15 08:48 PM, Matt Darnell wrote: > In the past we have used Adtran Atlas 550's to break out FXS ports for > devices like modems. The great thing about the 550 is that internally > it is all TDM so there is absolutely zero latency. > > We are able to use ATA's for faxes and analog phones but devices that > use modems, they fail 99.99% of the time when using an

I've had a (small) working asterisk with X100P and 4xFXS TDM card for a while now. I've been running the release from (I think) April 13 - asterisk 0.9.0, zap* 0.9.1. I tried 1.0 RC2, and my internal FXS line doesn't ring - call comes in the FXO, gets answered, and rings through to the FXS extension, but the phone doesn't ring - asterisk shows it ringing and eventually punts to