similar to: Echo during begining of incoming calls

Anyone tried the Clipcomm CG-410? I'm tired of beeing unable to get rid correctly of the echo problem. I have 3 TDM04B installed in one server. It was working fine when I had only one card installed in the server but since I installed 2 more then I can't get rid of it what ever I try... So now I'm looking at other solutions. And I saw that Clipcomm makes a router with 4 FXO ports

Hi I am interested in setting up an Asterisk PBX in my office with digium hardware, and I just have a few questions in regards to what I would need. It is my understanding that an FXO card is used to interface with an incoming/outgoing phone line, and an FXS card is used for interfacing with a phone within the system. Currently we have 4 incoming/outgoing phone lines and would like to have

I'm installing my first Asterisk server. I have a TDM04B card installed in my asterisk server (4x FXO ports). I have 5 Cisco IP phone 7960 working fine on asterisk using SIP. My configuration to receive call is working as expected meaning anyone calling on one of the 4 FXO ports is answer by asterisk and asked to enter the extension of the person to reach and then it is transfer on the

OK, here is the scenario, Asterisk @ Home 1.0 with TDM04B and TDM40B. I can receive and place calls with no issues, however, when I receive a call, the CID only shows "Analog Line" on the Grandstream 2000XP phone. Does anyone have any ideas even where to look to change this?? Is it a setting in the phone, Asterisk, or both?? Thanks, Marc

I have a customer running Asterisk 1.2.13, Zaptel 1.2.11 with a TE110P, a TDM04B and an Astribank-32. They have been complaining that there is echo on calls even when they are IP to IP on the same network. There are 18 Aastra 9133i phones and 30 analog phones connected to the Astribank. I can understand there being a bit of echo on the analog phones, but I do not understand why there would be

I have just installed Noteworthy Composer v2.0 It seems to work fine except when I try to call the print preview. The print preview shows up briefly and then Noteworthy shuts itself down. This is almost my first attempt to install anything using Wine. Not quite my first as I installed the Noteworthy Viewer as a trial and that worked once I had sorted out how to get MIDI working. Print preview

Hi everyone could somebody help me on this problem ??? I have a tdm04b card with 4 fxo's connected to 4 POTS of a media gateway. Suppose that I want to place the following calls: Zap/1 dials Zap/2 (by placing in /spool/outgoing a call file which dials a number corresponding to Zap/2) Zap/3 dials Zap/2 (also placing another call file) Zap/4 dials Zap/2 (also placing a call file) In each

Ok I know I'm not the only one having echo problem with asterisk but the weird thing is that when I receive a call from a PSTN line on my TDM04B card I don't have any echo problem at the beginning of the call then after a few minutes I start having echo on my side only (the person calling from a regular phone doesn't have any echo), then it stop and come back all the way until the

I just install a Digium TDM04B card. I created 4 separate Zap channels and one outbound routing containing zap channels from 1 to 4. If a phone line is plug in Zap/1 then works fine, but if I unplug phone line from Zap/1 (simulating fail) the system keep dialing out on Zap/1, even with no dial tone; Only if Zap/1 is busy it will use Zap/2. There is any work around or different setting to avoid

Hello, I have installed asterisk version 1.2.12 and latest zaptel modules. but i can see some IRQ conflicts on the server. iam uisng two TDM04B cards. according to my previous knowledge on asterisk verison 1.07 asterisk has given lot of erros when starting if you have assigned the same IRQ number to any other device. My question is new releae version 1.2.12 has resolved the IRQ issue ?? TDM04B

Hello all! We have an E400P on a dual Xeon box with four EuroISDN PRI to the PSTN and Cisco ATA-186 as SIP UA. We experience very large amount of echo (on -some- calls) when we're doing PRI->SIP calls but not when doing SIP->PRI calls. We don't think the problem is IP latency related (When calls are made PRI->*->PRI we still experience the echo). What do you think? Regards

Hello all! We have an E400P on a dual Xeon box with four EuroISDN PRI to the PSTN and Cisco ATA-186 as SIP UA. We experience very large amount of echo (on -some- calls) when we're doing PRI->SIP calls but not when doing SIP->PRI calls. We don't think the problem is IP latency related (When calls are made PRI->*->PRI we still experience the echo). What do you think?

Hello, I have installed asterisk version 1.2.12 and latest zaptel modules. but i can see some IRQ conflicts on the server. iam uisng two TDM04B cards. according to my previous knowledge on asterisk verison 1.07 asterisk has given lot of erros when starting if you have assigned the same IRQ number to any other device. My question is new releae version 1.2.12 has resolved the IRQ issue ?? TDM04B

Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13 Asterisk is being used as a meetme

OK I have been battling with echo problems with asterisk on ISDN for a few weeks now, and still can't solve it (although I think I have tried everything I can find.) I will try a post everything I think is possibly relevant that I can remember with the hope someone can point me in the right direction. current spec I am using (Trixbox 1.0 was still in beta at the time an didn't detect

-----Original Message----- My question concern outgoing calls. How can I configure my extensions.conf to get a PSTN line on my TDM04B card in the following order : first trying on the channel 4 then if 4 is busy then switch to 3 if 3 is busy then switch to 2 and if 2 is busy then say there's no more line available. I don't want to dial on the first channel as it's my main number

I have * installed as a voicemail only solution attached to a Toshiba Strata phone system. I am testing * at the current time, so it is not attached to ports designated as VM on the Toshiba. I am not sure if the VM ports would handle the call any different than an analog port. Currently if I call one of the DIDs that I have connected to the *, my greeting plays and I am able to dial my

I just wrote this little script that makes concurrent connections to a samba or NT server. It connects to a service and does a dir. This may be useful if you want to check the resources of your server. Configure it changing the vars at the begining. In my tests I compared two computers, NT vs samba-2.0.3 NT was much faster. 50 concurrent connections, 10 commands issued. NT-4.0 SP4 Ppro 166 ,

I am having a problem with a couple servers. They both have a Digium TE110P and a TDM04B card. I have setup the system so the TE110P uses channels 1-31 and the TDM04B 32-35. The problem is that when we reboot the server sometimes the TDM04B is recognized first and the TE110P second so the configuration fails. I do not know why this happens and to solve this I have to do a "service

In the next week or so, we'll be turning up a new T1 Pri from Cox Cable in Omaha using a T100P (installed but not yet configured). While discussing the interface parameters with a very knowledgable Cox engineer, we decided on some of the basics including b8zs, esf, dchan=24, callerid, 5 digits of called number forwarded to *, and switchtype=national (NI2). They indicated their CO switch is a