RE: [Asterisk-Users] Hi...Please help me

[Brian C. Fertig]

Chandra, 

 

In all honesty if they are proprietary and
you want to use them you will need a FXO card.  Alternatively there are

a few good termination providers out there
that are inexpensive. 

 

The top 3 most inexpensive that come to
mind are: 

 

Plainvoip  -   http://www.plainvoip.com Domestic
starting at 1.1c

VoipJet    -   http://www.voipjet.com    Domestic
starting at 1.3c

NuFone   -   http://www.nufone.net  Domestic
starting at 2c (I believe)

 

 

Anyone of these providers can supply you
with USA
and also international dialing.

 



_.._
Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE
Data/Telecom Engineer
IT Administrator
Planet Telecom, Inc
Tampa, FL
Office
o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 
SIP URI: [EMAIL PROTECTED]











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy
Sent: Monday, May 08, 2006 8:43 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Hi...Please help me



 

Hi Friends,

Thank you for your quick response. I have successfully implemented Intercom
(Dialling within my office LAN) using Asterisk. To implement this, I am using
X-Lite Softphone. 

Now, I want to make calls to US using VoIP Asterisk. 

I have registered with Vebtel (VoIP IP Telephony Service provider). They had
given me one VoIP Modem called "Voice Finder AP 200" and the below
values:

Inbound Number: 123456789
Public IP Number:
55.23.789.145
Password: xyz

(These values are dummy values)

Currently we are making US calls using VoIP provided by "Vebtel".
Now, I want to make US calls using this Vebtel service from Asterisk. How can I
do this?

I am unable to understand where to give above mentioned values? What
configuration files I should use to implement this using the Vebtel SIP
provider? Do I need to provide any more values to implement this using Asterisk
from Vebtel?

Waiting for your quick response. Thank you.  

Regards,
Chandra.







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PC-to-Phone Calls to the US
(and 30+ countries) for 2¢/min or less.





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[Asterisk-Users] Hi...Please help me

[Crazy Boy]

Hi Friends,  Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone.   Now, I want to make calls to US using VoIP Asterisk.   I have registered with Vebtel (VoIP IP Telephony Service provider). They had given me one VoIP Modem called "Voice Finder AP 200" and the below values:  Inbound Number: 123456789 Public IP Number: 55.23.789.145 Password: xyz  (These values are dummy values)  Currently we are making US calls using VoIP provided by "Vebtel". Now, I want to make US calls using this Vebtel service from Asterisk. How can I do this?  I am unable to understand where to give above mentioned values? What configuration files I should use to implement this using the Vebtel SIP provider? Do I need to provide any more values to implement this using Asterisk from Vebtel?  Waiting for your quick response. Thank you.    Regards, Chandra. 
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RE: [Asterisk-Users] Hi...Please help me

[William Piper]

Wouldn't it be easier to replace the callername to the exten.

example:

exten => _x.,1,SetCallerIDname(${EXTEN})
exten => _x.,2,SetCallerIDnum(${CALLERIDNUM})
exten => _x.,3,dial,SIP/number

That way, the Caller Name would show the extension it is ringing and the
callerid will still show the calling party.

Now you don't need the softphone to do it... just a phone with callerid.

bp

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, May 02, 2006 5:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Hi...Please help me

On Tuesday 02 May 2006 16:42, hugolivude wrote:
> 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.

That seems to be humongous overkill... why not just use any of the caller ID

popup apps instead of running that behemoth X-Lite?  If the popup comes up, 
the phone's for you.

-A.
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Re: [Asterisk-Users] Hi...Please help me

[Andrew Kohlsmith]

On Tuesday 02 May 2006 16:42, hugolivude wrote:
> 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…

That seems to be humongous overkill... why not just use any of the caller ID 
popup apps instead of running that behemoth X-Lite?  If the popup comes up, 
the phone's for you.

-A.
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Re: [Asterisk-Users] Hi...Please help me

[hugolivude]

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 syste

RE: [Asterisk-Users] Hi...Please help me

[William Piper]

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



 







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RE: [Asterisk-Users] Hi...Please help me

[John Joseph]

can u check what this command gives 
"iptables -L"

or do "iptables -F " [ Not advisable , but for testing
OK ]

 then try again 


--- Crazy Boy <[EMAIL PROTECTED]> wrote:

> 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 oughtto 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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>   
>
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RE: [Asterisk-Users] Hi...Please help me

[Crazy Boy]

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=tutorialadded 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 oughtto 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  
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RE: [Asterisk-Users] Hi...Please help me

[Evalyn Wafula]

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




From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy 
BoySent: 26 April 2006 14:56To: Asterisk Users Mailing 
List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 
Hi...Please help me
Hi,Thank you for your response. Basically, I follow "O Reilly 
AsteriskTFOT.pdf" book and some other eBooks. They have mentioned how to install 
Asterisk in server. But, they have not mentioned 

  What I have to install in client PC's?
  What hardware I need?
  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)
  Is it sufficient to buy hardware for server only OR for client PC's also?
  How can I connect my VoIP phone to server?
  How  can I connect hardware to server?
  How can I connect PSTN line to server PC?Please guide me to 
complete this task. Waiting for your response. Thank 
you.Regards,Chandra.Gonzalo Servat 
<[EMAIL PROTECTED]> wrote:
On 
  4/24/06, Crazy Boy <[EMAIL PROTECTED]>wrote:> Hi 
  Friends,>[..snip..]> ---> Employee 1 PC (Softphone 
  i.e., Headphones with Mic)> ---> Employee 2 PC (Softphone i.e., 
  Headphones with Mic)> ---> Employee 3 PC (Softphone i.e., 
  Headphones with Mic)> ---> --> ---> 
  --> ---> Employee 10 PC (Softphone i.e., Headphones 
  with> Mic)>> and vice versa.>> How can I 
  implement this? Is it possible to implement this using "Asterisk"> 
  software? If It can be implemented using "Asterisk" software, What 
  softwares> I should install in Server and Employee PC's? Is there any 
  need of buying> extra hardware?[..snip..]It can be done 
  with Asterisk. For the server side, you would need toinstall Asterisk on 
  your Fedora 5 box, Zaptel and lots of Wikireading.I don't 
  recommend using softphones for your employee PCs. It lookslike an 
  attractive solution at first (from a cost perspective) but inreality it's 
  not very practical (at least that was my experience).Buying 5 x 2 port 
  ATAs will cost you around $300-$350 which is notreally expensive 
  considering the kind of powerful PBX you will have atyour disposal. I 
  would have suggested some Digium hardware for the FXS(extensions) but I 
  think it will be a lot more expensive (for 10extensions) than the ATAs 
  solution. You could also look into a channelbank, but again it will be 
  more expensive than the 5 ATAs. As for theFXO (incoming/outgoing PSTN) I 
  recommend buying Digium hardware(TDM400P).Hope this helps, and 
  good 
  luck!Regards,Gonzalo.___--Bandwidth 
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Re: [Asterisk-Users] Hi...Please help me

[Crazy Boy]

Hi,Thank you for your response. Basically, I follow "O Reilly AsteriskTFOT.pdf" book and some other eBooks. They have mentioned how to install Asterisk in server. But, they have not mentioned What I have to install in client PC's?What hardware I need?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)Is it sufficient to buy hardware for server only OR for client PC's also?How can I connect my VoIP phone to server?How  can I connect hardware to server?How can I connect PSTN line to server PC?Please guide me to complete this task. Waiting for your response. Thank you.Regards,Chandra.Gonzalo Servat <[EMAIL PROTECTED]> wrote: On 4/24/06, Crazy Boy <[EMAIL PROTECTED]> wrote:> Hi Friends,>[..snip..]> ---> Employee 1 PC (Softphone i.e., Headphones with Mic)> ---> Employee 2 PC (Softphone i.e., Headphones with Mic)> ---> Employee 3 PC (Softphone i.e., Headphones with Mic)> --->--> --->--> ---> Employee 10 PC (Softphone i.e., Headphones with> Mic)>> and vice versa.>> How can I implement this? Is it possible to implement this using "Asterisk"> software? If It can be implemented using "Asterisk" software, What softwares> I should install in Server and Employee PC's? Is there any need of buying> extra hardware?[..snip..]It can be done with Asterisk. For the server side, you would need toinstall Asterisk on
 your Fedora 5 box, Zaptel and lots of Wikireading.I don't recommend using softphones for your employee PCs. It lookslike an attractive solution at first (from a cost perspective) but inreality it's not very practical (at least that was my experience).Buying 5 x 2 port ATAs will cost you around $300-$350 which is notreally expensive considering the kind of powerful PBX you will have atyour disposal. I would have suggested some Digium hardware for the FXS(extensions) but I think it will be a lot more expensive (for 10extensions) than the ATAs solution. You could also look into a channelbank, but again it will be more expensive than the 5 ATAs. As for theFXO (incoming/outgoing PSTN) I recommend buying Digium hardware(TDM400P).Hope this helps, and good luck!Regards,Gonzalo.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users
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Re: [Asterisk-Users] Hi...Please help me

[Gonzalo Servat]

On 4/24/06, Crazy Boy <[EMAIL PROTECTED]> wrote:
> Hi Friends,
>
[..snip..]
> ---> Employee 1 PC (Softphone i.e., Headphones with Mic)
> ---> Employee 2 PC (Softphone i.e., Headphones with Mic)
> ---> Employee 3 PC (Softphone i.e., Headphones with Mic)
> --->--
> --->--
> ---> Employee 10 PC (Softphone i.e., Headphones with
> Mic)
>
> and vice versa.
>
> How can I implement this? Is it possible to implement this using "Asterisk"
> software? If It can be implemented using "Asterisk" software, What softwares
> I should install in Server and Employee PC's? Is there any need of buying
> extra hardware?
[..snip..]

It can be done with Asterisk. For the server side, you would need to
install Asterisk on your Fedora 5 box, Zaptel and lots of Wiki
reading.

I don't recommend using softphones for your employee PCs. It looks
like an attractive solution at first (from a cost perspective) but in
reality it's not very practical (at least that was my experience).
Buying 5 x 2 port ATAs will cost you around $300-$350 which is not
really expensive considering the kind of powerful PBX you will have at
your disposal. I would have suggested some Digium hardware for the FXS
(extensions) but I think it will be a lot more expensive (for 10
extensions) than the ATAs solution. You could also look into a channel
bank, but again it will be more expensive than the 5 ATAs. As for the
FXO (incoming/outgoing PSTN) I recommend buying Digium hardware
(TDM400P).

Hope this helps, and good luck!

Regards,
Gonzalo.
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Re: [Asterisk-Users] Hi...Please help me

[Paul Hales]

It's all possible.

Paul Hales

--
Paul Hales
Technical Manager
Asterisk IT
bus: 03 8320 8100
mob: 0434 225 491

Crazy Boy wrote:


Hi Friends,

I want to implement VOIP PBX service in my office. I have 10 computers 
and a server. All computers are Pentium IV processors with 512 MB RAM. 
All employee computers have Windows 2000 Professional OS and Server 
computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have 
a VOIP phone and have registered with VoIP service provider. Now, I 
want to implement VOIP PBX facility to all of my systems.


The structure for the same is:

PSTN (Phone call) ---> VOIP phone ---> Server system --->

---> Employee 1 PC (Softphone i.e., Headphones 
with Mic)
---> Employee 2 PC (Softphone i.e., Headphones 
with Mic)
---> Employee 3 PC (Softphone i.e., Headphones 
with Mic)

--->--
--->--
---> Employee 10 PC (Softphone i.e., Headphones 
with Mic)


and vice versa.

How can I implement this? Is it possible to implement this using 
"Asterisk" software? If It can be implemented using "Asterisk" 
software, What softwares I should install in Server and Employee PC's? 
Is there any need of buying extra hardware?


Please reply me. Thank you

Thanks & Regards,

Chandra.


Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great 
rates starting at 1¢/min. 
 







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Re: [Asterisk-Users] Hi...Please help me

[Marcel Hecko]

For hardware check out this page:
http://www.digium.com/en/products/hardware/

Marcel

Crazy Boy wrote:
> Hi Friends,
> 
> I want to implement VOIP PBX service in my office. I have 10 computers
> and a server. All computers are Pentium IV processors with 512 MB RAM.
> All employee computers have Windows 2000 Professional OS and Server
> computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a
> VOIP phone and have registered with VoIP service provider. Now, I want
> to implement VOIP PBX facility to all of my systems.
> 
> The structure for the same is:
> 
> PSTN (Phone call) ---> VOIP phone ---> Server system --->
> 
> ---> Employee 1 PC (Softphone i.e., Headphones with Mic)
> ---> Employee 2 PC (Softphone i.e., Headphones with Mic)
> ---> Employee 3 PC (Softphone i.e., Headphones with Mic)
> --->--
> --->--
> ---> Employee 10 PC (Softphone i.e., Headphones with
> Mic)
> 
> and vice versa.
> 
> How can I implement this? Is it possible to implement this using
> "Asterisk" software? If It can be implemented using "Asterisk" software,
> What softwares I should install in Server and Employee PC's? Is there
> any need of buying extra hardware?
> 
> Please reply me. Thank you
> 
> Thanks & Regards,
> 
> Chandra.
> 
> Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great
> rates starting at 1¢/min.
> 
> 
> 
> 
> 
> ___
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> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Hi...Please help me

[Marcel Hecko]

Yes it is possible - check out the Asterisk manual or nice book from
O'Reilly - Asterisk PBX (The Furute of telephony)

Marcel

Crazy Boy wrote:
> Hi Friends,
> 
> I want to implement VOIP PBX service in my office. I have 10 computers
> and a server. All computers are Pentium IV processors with 512 MB RAM.
> All employee computers have Windows 2000 Professional OS and Server
> computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a
> VOIP phone and have registered with VoIP service provider. Now, I want
> to implement VOIP PBX facility to all of my systems.
> 
> The structure for the same is:
> 
> PSTN (Phone call) ---> VOIP phone ---> Server system --->
> 
> ---> Employee 1 PC (Softphone i.e., Headphones with Mic)
> ---> Employee 2 PC (Softphone i.e., Headphones with Mic)
> ---> Employee 3 PC (Softphone i.e., Headphones with Mic)
> --->--
> --->--
> ---> Employee 10 PC (Softphone i.e., Headphones with
> Mic)
> 
> and vice versa.
> 
> How can I implement this? Is it possible to implement this using
> "Asterisk" software? If It can be implemented using "Asterisk" software,
> What softwares I should install in Server and Employee PC's? Is there
> any need of buying extra hardware?
> 
> Please reply me. Thank you
> 
> Thanks & Regards,
> 
> Chandra.
> 
> Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great
> rates starting at 1¢/min.
> 
> 
> 
> 
> 
> ___
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> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Hi...Please help me

[Crazy Boy]

Hi Friends,I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP phone and have registered with VoIP service provider. Now, I want to implement VOIP PBX facility to all of my systems. The structure for the same is:PSTN (Phone call) ---> VOIP phone ---> Server system --->             ---> Employee 1 PC (Softphone i.e., Headphones with Mic)            ---> Employee 2 PC (Softphone i.e., Headphones with Mic)            ---> Employee 3 PC (Softphone i.e., Headphones with
 Mic)                    --->     --                    --->     --            ---> Employee 10 PC (Softphone i.e., Headphones with Mic)and vice versa.How can I implement this? Is it possible to implement this using "Asterisk" software? If It can be implemented using "Asterisk" software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware? Please reply me. Thank youThanks & Regards,Chandra.
		Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls.  Great rates starting at 1¢/min.___
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