similar to: sip and x-lite

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'm just finished installing the TDM (2 port) and X100P. I'm using X100P to pstn, and the TDM to the phone. I've loaded the module, and I can also list the card in the /proc/zaptel/ I'm a little confused now. in zapatel.conf, how do I know which channel is which. (TDM or X100P)? Thanks and pardon my English Isianto Istiadi

This newbie has been trying out Asterisk. It has been both a) surprisingly painful and b) impressive in terms of helpful support from other users. Having got two phones to communicate and then got voicemail MWI going (neither painlessly) I decided the next step was to implement call transfer as per nearly all commercial PBX systems i.e. hold call consult another extension either exit and let

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

The cheap ones on EBay won't work with the SC420 server. I have one and can't make any of the clones work. I do have one TDM40B card for analog stations that works well. The problem with the SC420 is that it won't let you set the interrupts yourself and you end up with interrupts being shared. =============================================================== Message: 26 Date:

Hi All, Our experience with * to date has been a bit limited. It's a 4xCisco 7960 network, linking our head office with a faraday caged datacenter. As a way of putting voicecomms into a sealed room, it was quick and easy to deploy, and works very well. As typically happens, we've now thought about extending the use of asterisk - and a new opportunity has cropped up. In three months

To be clear I meant using Chan)_h323 with Call Manager where CM is configured with * as a H.323 gateway, not client. CM supports H.323 to direct calls through gateways, and in fact until recently that is all they supported. They now also have MGCP, but only to their IOS platforms, and SIP is coming soon. There are NO sccp-based gateways, from Cisco anyways. Dan -----Original Message-----

> 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.

Hi, this code used to work, but for some reason it won''t now: [... load models] DataMapper.setup(:default, "mysql://cm:password-yro0w6Hh2UrkfKl135yGWg@public.gmane.org/cm") DataMapper.setup(:lite, "sqlite3:///home/tigre/cm.sqlite") DataMapper.finalize DataMapper.repository(:lite).auto_migrate! DataMapper.repository(:lite) do w = WorkCenter.new(:name =>

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

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

Hi folks! I bought a X100P a while ago and know I've tried to get it working here at home again ... but I can't manage to get my X-Lite client working with Asterisk (CVS from a day ago) ... I've downloaded the latest version of X-Lite and I believe that I've set it up correctly ;-) But I cant get it to register with my Asterisk - I only get "Login timed out, contact your

Hi, we are running Samba 2.2.5 using LDAP und pam_ldap (pam_unix2 with auth+account+password=use_ldap) as PDC out of the SuSE 8.1 distribution. It runs very well: Login f?r Unix&Samba ok, Passwort-Change for Samba via smbpasswd Ok and we are able to manipulate the Linux Password in LDAP using the GQ Client. The only thing that doesn't work is "passwd" itself: venezuela:/home/tdm

Try the snom soft phone! http://snom.com CS > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Dave Chase > Sent: Saturday, February 26, 2005 12:31 PM > To: ich@mateo.ch; Asterisk Users Mailing List - > Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Anybody using

Gentlemen, Forgive me if I am posting at the wrong place! I was going to test the "new" chan_ooh323 driver so I did install: debian: Linux sip2 2.6.26-2-686 #1 SMP dahdi-linux-complete-2.2.0.2+2.2.0 Asterisk SVN-trunk-r231692 Did enable chan_ooh323, everything compiled without any problems. Hardware setup: Phone (975) - Avaya CM - H.323 - Asterisk - X-Lite (0317998975) X-Lite can

Hello People, I am a newbie asterisk and happy user, i have configured a x100p card and everything works nice, i can forward incoming connections to a x-lite software client and works out of the box, However when i try to make a connection between two x-lite clients then no audio is transmited, i have followed the instructions on voip-info.org, the tutorials on onlamp and i have read some

Mateo, Dialing the extension to your softphone is the same as any hardware extension. Exten => 1000,1,Dial,(SIP/1000,20,trf) pretty exten => 1000,2,Macro(vmessage,1000) exten => 1000,3,Hangup Change [mateo01] to [1000] in your sip and you will be saying that ext. 1000 is registered with the specifics you are using. Update the settings in your softphone to register the name and

XLite does not support transfer... You have to buy their XPro -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mateo Meier Sent: Tuesday, February 22, 2005 3:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Anybody using X-Lite Softphone ? tryed to forwarda call to X-Lite.... Hey Guys Im

That's the main reason I opened this thread as it surprised me a bit ... > > > Any 2-wire analog leg will be a source of echo. Many, many, many calls > do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? > Think cell/mobile or endpoints with PRI or T-1. > >