similar to: Newbe Questions.

you can do that. But are u installing qmail and * on same box. i wont recommend that. i use qmail and *. qmail is strictly for internet email. * is on separate server not exposed to Internet. * box also has sendmail. i hv configured sendmail to use smart host (qmail server). This way its safe and secure. HTH, -B ----- Original Message ----- From: "Ing Isianto Istiadi"

Dear all, I have some questions, I'm sure it's pretty stupid for most of you, but I need you guys to help me. Here are my questions: 1. Music On Hold, it doesn't play any sound on the parked call or hold call. But if I do ps-ax, it shows mpg123 .....( I forgot the exact line). I'm using slackware 9.1 2. I have fxs 3 port, and in my zapata.conf I have included callpickup=1-4,

Dear All, I''m using the kernel 2.6.6, iproute2-2.4.7.20020116, iptables v1.2.9, and gentoo. I have a leased-line 64 kbps. I can see the counter works in iptables, but in the htb, it doesn''t go to the right class (it always go to the default class). Any help will be appreciated here''s my htb conf #!/bin/bash tc qdisc del dev eth1 root tc qdisc add dev eth1 root

> 3. Supposed I have 2 fxo cards (right now I have one already) and 3 > fxs, and one of the fxo will have two phone (running pararell), is > there any way for * to: > a. It always dial the first fxo, if the fxo is busy or is being used > (have other people conversation), will * be able to switch it to > other fxo? Here's the approximiate the conditions of the phone.

Hello All, I loaded Asterisk@Home. I'm using SLPhones and can connect to mailboxs on the system. I have one X100P card. I try to dial out but get rejected. Any help... Thanks, David

Just to make sure this isn't a typo in your original email... Is this example from your zapata.conf? Also, the extension you have shown are in extensions.conf not zapata.conf correct? Here is an example of a good zapata.conf.... [channels] language=en busydetect=yes faxdetect=both busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes useincomingcalleridonzaptransfer=yes

try this... http://www.fnords.org/~eric/asterisk/ cm Thanks for the Info, and It worked. But I have a couple of questions: 1. There's an echo. How to get rid of the echo? 2. Is there any way to call from x-lite just the extention number? (say that in my extention.conf, I have extention 32 to connect to my fxs card (TDM). If I just call 32, it will time out. The work around that I did is to

I have a * box running * version 1.0.3 with two X100P line cards in it and Cisco 7960 IP phones. Everything seems to work pretty well with the exeption that the system hangs up on phone calls for no apparent reason. It does this on both incoming and outgoing calls through the POTS line (currently only have one). The only thing in the asterisk console with maximum verbousity is " -- Hungup

I have two grandstream budtetone-100 and cisco 7960g phones. When I talk via speaker phone on either of the phones I get a lot of echo. Any suggestions? Also how do I turn on the mark echo canceller. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, April 13,

Hi list. I'm having the next problem. I have a * with 1 TDM400P (4 ports) and one X100P, with a working configuration. Today i add one more X100P card, and i change the config files as next: zapatel.conf: fxsks=1-2 fxoks=3-6 loadzone = us defaultzone=us zapata.conf ... context=bell

Dear all, Is * capable to use qmail as a MTA? If so, how can I set it? I'm using asterisk v0.5, and TDM30B (FXS), Wildcard X100P(FXO), and x-lite(SIP softphone). In zapata.conf, I put already callwaiting=yes. My PSTN doesn't not support the callwaiting feature, so I don't expect the FXO is call waiting enabled. The question is can FXS and SIP support call waiting?? Cause everytime I

This may seem obvious or silly but have you tried using a different phone cord? A bad phone cord played havoc on a colleague of mine during initial config. Fro what you show as your output from ztcfg, you should have one channel configured successfully then. Your example shows you have the channel set to one so no problem there. If you run ZAP SHOW CHANNELS from the CLI what do you see? W

You should see something like this.... (I have 8 channels) tuxpbx*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo incoming en 1 incoming en 2 incoming en 3 incoming en 4 incoming en 5 incoming en 6

Hello! I've been playing with two pieces of hardware: a X100P and a TDM400P with an FXO and two FXS modules. I had been using just the TDM card; however, the TDM FXO module seems to hear things and "answer" the telephone for no reason, and I wanted to compare the results with an X100P card. If you want further details, I can give them to you, but suffice it to say that

Dear all, I have a question regarding the configuration of *. I have 3 port FXS, and 2 port FXO. I have 4 users that use analog phone connected to FXS (I have 3 phones). I need to limit the user's capability (user A can call International, user B can call long distance, etc). I want to implement the password say to call , he/she needs to puch 9(for outgoing call)2-4 digits password,then

Good news! FYI, worked with Mark this afternoon to test changes needed to reduce or eliminate echo involving pstn calls on the new tdm fxo card (bug #1902). Three simple source code changes (testing-only at this point) resulted in zero detectable echo on all incoming and outgoing tdm pstn calls, even in the first second or two. I've not noticed any unusual side effects at all. Since the

*** Now you just need a "phone" or "station" (FXS) device and a method of connecting it. *** So... essentially I need to spend another $125 US on a Wildcard TDM400P so I can plug in an analog phone? On the X100P description it says: "The X100P supports FXS Loopstart and "Kewlstart" (Loopstart with far end disconnection supervision). It can detect ringing and

For reference... http://www.voip-info.org/wiki-Asterisk+zap+channels Not sure it is relevant but go ahead and remove the spacing on the channel line so it will read.... channel=>1 Here is the original incoming context you showed. [incoming] exten => s,1,Answer ; Answer the line exten => s,2,Playback,demo-thanks ;for playing a file The Playback looks malformed based upon the wiki

In having configured my first * server there are a few questions I could not understand or find answers to 1) How does one use ztmonitor to adjust the rxgain and txgain. I have set mine to -1.0 each to get rid of echo on std phones connected on the TDM10B FXS module 2) Is it best to use a TDM card with an TFX and FXO module or is the the FXO = X100p and FXS on TDM a good method? 3) How do I

Hey all, I recently noticed that DTMF caller-ID was implemented in CVS, so I requested the service from my telco (the Dutch KPN) and tried to get it going in Asterisk (current CVS), without success so far. This system has 1 X100P, 2 TDM400P's with 4 FXS-modules each and 2 HFC-PCI ISDN-cards (zaphfc-driver) in it. The analog line I'm trying to get caller-ID working on is obviously on the