First off, I agree w/ Gonzalo – softphones didn't work out for me
either. One thing that did work great tho was a combo.
We share SIP phones at the office in a 1:4 ratio. You're probably
asking – how do you know when a ringing phone is for you? Well,
everyone in our office gets an XLite softphone, and I direct calls to
make BOTH the SIP phone AND the XLite ring. If your XLite pops up,
you know that ring phone is for you…
Here's some answers to your other questions
• What I have to install in client PC's?
Just the softphone client (e.g. XLite (SIP) Cubix (IAX)
http://www.virbiage.com/cubix.php
• What hardware I need?
Nothing too fancy. Your PCs seem OK. For Asterisk, I'm using an old
Pentium 4 beater with 1Gig memory and it handles the whole office (19)
just fine.
• How can I take decission to buy extra hardware (like Zaptel
products) OR no need of buying extra hardware? ( I will be using
Asterisk for 70 PC's and a server)
This depends on what you want in the way of handsets, and what kind of
connectivity you want to the PSTN (Public Switched Telephone Network).
You could get away with no extra hardware in a pure VoIP solution.
Connect Asterisk to the Internet w/ an Ethernet cable and use SIP
based phones that also communicate over a network. Note that if you
don't use any Digium hardware, I believe that you need to use ztdummy
to control timing (never used it myself)
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
• Is it sufficient to buy hardware for server only OR for client PC's
also?
Again, your PCs seem OK. How you kit out your server depends upon
what you want.
• How can I connect my VoIP phone to server?
Once you have Asterisk installed, you have to configure your VoIP
phone to register with it. For example, look here for how to
configure Polycom Soundpoint 501s -
http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501.
You'll also have to have the appropriate entries in SIP.conf for the
phone AND to connect to your VoIP service provider
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
• How can I connect hardware to server?
Don't understand this one. If you use telephony boards, you'll need
drivers. Depending upon the board you may also have to physically
connect your phone to it with a telephone wire (as is the case with
TDM boards for example)
• How can I connect PSTN line to server PC?
Assuming analogue phones you'll need a TDM card with an FXO port
(outgoing) for each line you have
(http://www.digium.com/en/products/hardware/analogcards.php). You'll
also need an FXS port for each phone you have on your TDM card as
well.
Yours,
H
On 5/2/06, William Piper <[EMAIL PROTECTED]> wrote:
You are missing the dtmf mode, and most importantly… the codec to be used.
I would also add the nat=yes, that is probably why your phone isn't
registering.
See below for example config:
[chandra]
type=friend
username=chandra
secret=chandra
nat=yes
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=g729
context=tutorial
canreinvite=no
________________________________
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Crazy Boy
Sent: Tuesday, May 02, 2006 8:58 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Hi...Please help me
Hi friends,
Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6
version. I have installed Asterisk in my PC and "X-Lite" as softphone in my
PC and client PC. Here my user name is "chandra" and client user name is
"aarti". I have added these lines to configuration files at the end of file.
added contents in sip.conf:
[aarti]
type=friend
username=aarti
secret=aarti
host=dynamic
context=tutorial
[chandra]
type=friend
username=chandra
secret=chandra
host=dynamic
context=tutorial
added contents in extensions.conf:
[tutorial]
exten => 101,1,Dial(SIP/aarti)
exten => 102,1,Dial(SIP/chandra)
Here, "aarti" is client, "chandra" is mine and Asterisk is also installed
in my PC (chandra) and it is successfully connected to Asterisk server using
"X-Lite" softphone.
But, when i try to connect from "aarti" system using softphone, it displays
an error message "login timedout, contact system admin".
Is there any problem with the content of sip.conf file or extensions.conf
file? I have not connected any external hardware to my pc. I just want to
connect Asterisk server to my collegues PC's like Intercom within my office
LAN using headphones. How can I do this? Please tell me. Looking forward for
your response.
Thank you.
Regards,
Chandra.
Evalyn Wafula <[EMAIL PROTECTED]> wrote:
Hi Chandra, I am also new to Asterisk and I have only just started
installing a test system but I probably can help clarify one or two things.
I think asterisk "clients" are phones not PCs unless you use "soft phones"
which is software on the PC (somewhat like Skype) that you use to make and
answer phone calls. So you might not need to install anything on your PCs if
you will use IP phones or ATAs as mentioned by Gonzalo.
The hardware you need depends on what you require your asterisk to do. If
you will be making only IP calls using IP phones, then you only need
asterisk running on your server with no extra hardware. But if you need to
connect with analog/digital phone equipment, then you need extra hardware on
the server.
You do not physically connect your VOIP phone to the asterisk server. You
connect it to the network that has the server through a normal network point
and configure it to find the server.
You probably ought to take Gonzalo's advice and head over to:
http://www.voip-info.org/wiki-Asterisk and do some reading
before you even start as it will help you fit many pieces of the asterisk
"puzzle" together. It helped me get started. Then you probably will have
fewer questions that list members will answer more readily :)
Regards
Wafula
________________________________
Blab-away for as little as 1¢/min. Make PC-to-Phone Calls using Yahoo!
Messenger with Voice.
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