search for: isianto

...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" <isianto.istiadi@adirarental.com> To: <asterisk-users@lists.digium.com> Sent: Thursday, January 15, 2004 12:09 AM Subject: [Asterisk-Users] Sending voicemail with qmail > > Dear all, > Is * capable to use qmail as a MTA? > If so, how can I set it? > Thanks...

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

I'm pretty much a newbie to this but still think I've been around the various help pages, voip-info.org etc to be fairly sure I'm not missing something here so your help is appreciated! I have a box running RedHat9 at home with the latest CVS of Asterisk and all works fine. At the office, we installed Gentoo linux on a machine, downloaded the latest CVS of Asterisk, set it up. All

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

...alling 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

How do I get Gnophone to register to my Asterisk server? I have set up iax. conf as follows: [tim] type=friend ;username=tstornes host=dynamic ;defaultip=207.194.60.56 secret=1111 context=from-iax callerid => "Tim" <5000> auth=plaintext qualify=10 permit=0.0.0.0/0.0.0.0 and extensions.conf includes a section in the context from-iax: exten => 5000,1,Dial(IAX/tim/s|100|r)

...ion.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 add the user that I want to call to phone book with the extention like 32@192.168.1.1, is there any way to do this? Did I miss something in the configurations? Thanks Isianto 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 called my FXS, it always gave me a busy signal (when the fxs port is in use). Thanks Isianto

Hy guys, I cannot understand How the asterisk works. I would like know how the h323.conf, sip.conf and extension.conf works. I don't understand the parameters and the [sections]. What I need to the asterisk get a SIP call and forward them to a H323 terminal. I working at the h323.conf and extension.conf but I cannot understand!!! Please someone can help me. I your can send me a example (with

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

...r automatically, say that my friend call me to *, but nobody pick it up, can it automatically forward the call to my cellphone number? If it can, can you show me how to do it? 3. for music on hold do I need to put something in my extension.conf or it will work out of the box? Thanks in advance Isianto Istiadi

I am trying to build asterisk-0.9.0 on 533MHz 160MB Redhat Linux 9.0 machine. I have followed the instructions to build asterisk. Building zaptel and libpri seemed to go well (lots of messages but nothing that indicated an error) However, when I do the make clean ; make install for asterisk-0.9.0 after running for sometime I get the following: gcc -shared -Xlinker -x -o app_senddtmf.so

...sers@lists.digium.com > Subject: Re: [Asterisk-Users] user password and call waiting > Reply-To: asterisk-users@lists.digium.com > > Use account codes. That works ALOT better. If you require passwords then > look at app_authenticate. > > bkw > > On Tue, 20 Jan 2004, Ing Isianto Istiadi wrote: > > > > > Dear all, > > I have a questions: > > 1. I have 3 FXO connected to 3 analog phones. But I have 5 users using those > > phone. I want to be able to log who is using the phones and where to. I'd > > like to use password for each user...