asterisk users - Jan 2004 - Newbie Question-Looking for Feedback

I've been looking at Asterisk for a replacement for our phone system and
I'm
hoping someone can help validate my assumptions.

We have 4 analog lines coming into the building. These lines are simple POT
lines and we have them in a "hunt group" with Verizon so that when a
single
phone number is dialed, the first line is rang, if that line is busy it will
ring the second line, and so on.

I would like to put together an Asterisk system to handle these lines and
allow us to do VoIP, call queuing, voice menuing, etc. In looking at the
product offerings of Digium, it appears that I need 4 Wildcard X100P's and 1
Wildcard TDM400P 4-Port. For VoIP work, I'm looking for any recommendations
that can be made. My first priority is to support a user in New Zealand
talking to our phone system in the US, but there could be another 2 I'd like
to support in the US (all on cable modems with the typical capped 30KB/s
upload). I'd like for it to work very well with the Asterisk PBX and be as
simple as plugging in a ethernet cable or even support 802.11b/g with little
to no configuration.

In addition, I'm curious on other people's experience with software
based
VoIP phone. Specifically, it appears that a good amount of amount could be
saved by using software based phones inside the building, thereby negating
the need to purchase 3 hard VoIP phones and the Wildcard TDM400P. Can anyone
recommend a good software package, that is fairly idiot proof and would work
well for a small call-center with temp/minwage employees?


To Summarize:
- Can and does it make sense to purchase 4 Wildcard X100P's?
- Can and does it make sense to purchase 1 Wildcard TDM400P (4-Port)?
- What VoIP hard phone works best with Asterisk? Are there WiFi ones that
are less than $100?
- How much bandwidth does VoIP require? Will cable modem users with a max
30KB/s upload ok?
- What VoIP soft phone works best with Asterisk?

Also:
- What kind of uptime are people experiencing?
- How much system load will be needed for 4 concurrent VoIP conversations?
- What kind of gotcha's have people had that would be good for a newbie to
know?


Thank you in advance,

Matt

M. Matt Colgin wrote:
>I've been looking at Asterisk for a replacement for our phone system and
I'm
>hoping someone can help validate my assumptions.
>
I'll try.. :)
>
>We have 4 analog lines coming into the building. These lines are simple POT
>lines and we have them in a "hunt group" with Verizon so that when
a single
>phone number is dialed, the first line is rang, if that line is busy it will
>ring the second line, and so on.
>
>I would like to put together an Asterisk system to handle these lines and
>allow us to do VoIP, call queuing, voice menuing, etc. In looking at the
>product offerings of Digium, it appears that I need 4 Wildcard X100P's
and 1
>Wildcard TDM400P 4-Port. For VoIP work, I'm looking for any
recommendations
>that can be made. My first priority is to support a user in New Zealand
>talking to our phone system in the US, but there could be another 2 I'd
like
>to support in the US (all on cable modems with the typical capped 30KB/s
>upload). I'd like for it to work very well with the Asterisk PBX and be
as
>simple as plugging in a ethernet cable or even support 802.11b/g with little
>to no configuration.
>
>In addition, I'm curious on other people's experience with software
based
>VoIP phone. Specifically, it appears that a good amount of amount could be
>saved by using software based phones inside the building, thereby negating
>the need to purchase 3 hard VoIP phones and the Wildcard TDM400P. Can anyone
>recommend a good software package, that is fairly idiot proof and would work
>well for a small call-center with temp/minwage employees?
>
>
>To Summarize:
>- Can and does it make sense to purchase 4 Wildcard X100P's?
>- Can and does it make sense to purchase 1 Wildcard TDM400P (4-Port)?
>
I will answer these together, the recomendation is typically not to go 
above 3 cards in a system which means that you could give 5 cards a go 
but chances are you are not going to have a happy time with it..

My suggestion would be to either use a channelbank and a T100P or the 
simpler solution convert your 4 analog lines to 2 ISDB BRI lines and 
then get a 2 port AVM or Eicon ISDN card..
>- What VoIP hard phone works best with Asterisk? Are there WiFi ones that
>are less than $100?
>
My personal favorite in terms of both cost and performance would have to 
be the Snom 200.. Other options are the Grandstream (cheapest there is), 
the Cisco(a little pricey), the Snom 105 and no doubt a few others..

A Grandstream costs about $75 and AFAIK its still the cheapest so I 
would have to say No, you will not likely get a WiFi VoIP phone for 
under $100..
>- How much bandwidth does VoIP require? Will cable modem users with a max
>30KB/s upload ok?
>
The bandwidth requirement is dependent on the codec but 30KB/s should 
hande any codek no problem.. the bigger problem you will have between NZ 
and the US is latency which is really annoying when trying to transfer 
realtime data..
>- What VoIP soft phone works best with Asterisk?
>
I have found X-Lite or X-Pro to be the best..
>
>Also:
>- What kind of uptime are people experiencing?
>
I have over 100 days continuous, with reboots to apply patches.. Others 
on the list have said thay have over a years uptime..
>- How much system load will be needed for 4 concurrent VoIP conversations?
>
I have a P2 400 development server and have done 4 concurrent ( thats 
all I have ) VoIP sessions.. My production server is more powerful and i 
have not really looked at the number of concurrent sessions but its 
never really broken a sweat..
>- What kind of gotcha's have people had that would be good for a newbie
to
>know?
>
Becasue the software is free does not mean that the system will be, 
there are many really cheap off the shelf analog only PBX's out there 
that will be much cheaper.. the advantage to * is all the features you 
get and the VoIP support.. and of course the satisfaction when its all 
working.. Of course if it crashes you better run cos your users will be 
hunting you down.. :)
>
>
>Thank you in advance,
>  
>
Hope it helped..

Later..

Greetings all. I am new to the Asterisk world! Found it very impressive so far!

In relation to the below.

I have worked with Alcatel PBX's for the last 3 years. Alcatel OxE supports
SIP and H323 as well.
As far as SIP goes I have also found the Xlite to be good for soft phones. I am
using one now.
Check out www.xten.com Xlite is free and easy to use. I also have been given a
Pingtel SIP to play with. http://www.pingtel.com/
As far as H323 terminals go I have not played with all that many, however the
simple Microsoft netmeeting works for testing purpose anyway.

Now a question to all you experts out there, and this may seem VERY stupid, but
I have configured the sip phone and have it logged in and can dial 500 to get to
the sample messages etc. However i cannot work out how to give the sip termainal
a number that can be dialled. I would assume that it needs to be in the
dialplan, so I have added it in via the extensions.conf file, however I am sure
that I have stuffed the config somewhere. Can someone please point me in the
right direction. Would be much appreciated. Also, do i need hardware to make a
SIP to SIP call... eg. Compressors etc.

Cheers

Chris


-----Original Message-----
From: WipeOut [mailto:wipe_out@users.sourceforge.net]
Sent: Thursday, 8 January 2004 8:32 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Newbie Question-Looking for Feedback


M. Matt Colgin wrote:
>I've been looking at Asterisk for a replacement for our phone system and
I'm
>hoping someone can help validate my assumptions.
>
I'll try.. :)
>
>We have 4 analog lines coming into the building. These lines are simple POT
>lines and we have them in a "hunt group" with Verizon so that when
a single
>phone number is dialed, the first line is rang, if that line is busy it will
>ring the second line, and so on.
>
>I would like to put together an Asterisk system to handle these lines and
>allow us to do VoIP, call queuing, voice menuing, etc. In looking at the
>product offerings of Digium, it appears that I need 4 Wildcard X100P's
and 1
>Wildcard TDM400P 4-Port. For VoIP work, I'm looking for any
recommendations
>that can be made. My first priority is to support a user in New Zealand
>talking to our phone system in the US, but there could be another 2 I'd
like
>to support in the US (all on cable modems with the typical capped 30KB/s
>upload). I'd like for it to work very well with the Asterisk PBX and be
as
>simple as plugging in a ethernet cable or even support 802.11b/g with little
>to no configuration.
>
>In addition, I'm curious on other people's experience with software
based
>VoIP phone. Specifically, it appears that a good amount of amount could be
>saved by using software based phones inside the building, thereby negating
>the need to purchase 3 hard VoIP phones and the Wildcard TDM400P. Can anyone
>recommend a good software package, that is fairly idiot proof and would work
>well for a small call-center with temp/minwage employees?
>
>
>To Summarize:
>- Can and does it make sense to purchase 4 Wildcard X100P's?
>- Can and does it make sense to purchase 1 Wildcard TDM400P (4-Port)?
>
I will answer these together, the recomendation is typically not to go 
above 3 cards in a system which means that you could give 5 cards a go 
but chances are you are not going to have a happy time with it..

My suggestion would be to either use a channelbank and a T100P or the 
simpler solution convert your 4 analog lines to 2 ISDB BRI lines and 
then get a 2 port AVM or Eicon ISDN card..
>- What VoIP hard phone works best with Asterisk? Are there WiFi ones that
>are less than $100?
>
My personal favorite in terms of both cost and performance would have to 
be the Snom 200.. Other options are the Grandstream (cheapest there is), 
the Cisco(a little pricey), the Snom 105 and no doubt a few others..

A Grandstream costs about $75 and AFAIK its still the cheapest so I 
would have to say No, you will not likely get a WiFi VoIP phone for 
under $100..
>- How much bandwidth does VoIP require? Will cable modem users with a max
>30KB/s upload ok?
>
The bandwidth requirement is dependent on the codec but 30KB/s should 
hande any codek no problem.. the bigger problem you will have between NZ 
and the US is latency which is really annoying when trying to transfer 
realtime data..
>- What VoIP soft phone works best with Asterisk?
>
I have found X-Lite or X-Pro to be the best..
>
>Also:
>- What kind of uptime are people experiencing?
>
I have over 100 days continuous, with reboots to apply patches.. Others 
on the list have said thay have over a years uptime..
>- How much system load will be needed for 4 concurrent VoIP conversations?
>
I have a P2 400 development server and have done 4 concurrent ( thats 
all I have ) VoIP sessions.. My production server is more powerful and i 
have not really looked at the number of concurrent sessions but its 
never really broken a sweat..
>- What kind of gotcha's have people had that would be good for a newbie
to
>know?
>
Becasue the software is free does not mean that the system will be, 
there are many really cheap off the shelf analog only PBX's out there 
that will be much cheaper.. the advantage to * is all the features you 
get and the VoIP support.. and of course the satisfaction when its all 
working.. Of course if it crashes you better run cos your users will be 
hunting you down.. :)
>
>
>Thank you in advance,
>  
>
Hope it helped..

Later..

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users

Back in December there was a thread or remark about a Digium quad FXO 
card.  I would like to know when Digium will start marketing this...

M. Matt Colgin wrote:
> I've been looking at Asterisk for a replacement for our phone system
and I'm
> hoping someone can help validate my assumptions.
> 
> We have 4 analog lines coming into the building. These lines are simple POT
> lines and we have them in a "hunt group" with Verizon so that
when a single
> phone number is dialed, the first line is rang, if that line is busy it
will
> ring the second line, and so on.
> 
> I would like to put together an Asterisk system to handle these lines and
> allow us to do VoIP, call queuing, voice menuing, etc. In looking at the
> product offerings of Digium, it appears that I need 4 Wildcard X100P's
and 1
> Wildcard TDM400P 4-Port. For VoIP work, I'm looking for any
recommendations
> that can be made. My first priority is to support a user in New Zealand
> talking to our phone system in the US, but there could be another 2 I'd
like
> to support in the US (all on cable modems with the typical capped 30KB/s
> upload). I'd like for it to work very well with the Asterisk PBX and be
as
> simple as plugging in a ethernet cable or even support 802.11b/g with
little
> to no configuration.
> 
> In addition, I'm curious on other people's experience with software
based
> VoIP phone. Specifically, it appears that a good amount of amount could be
> saved by using software based phones inside the building, thereby negating
> the need to purchase 3 hard VoIP phones and the Wildcard TDM400P. Can
anyone
> recommend a good software package, that is fairly idiot proof and would
work
> well for a small call-center with temp/minwage employees?
> 
> 
> To Summarize:
> - Can and does it make sense to purchase 4 Wildcard X100P's?
> - Can and does it make sense to purchase 1 Wildcard TDM400P (4-Port)?
> - What VoIP hard phone works best with Asterisk? Are there WiFi ones that
> are less than $100?
> - How much bandwidth does VoIP require? Will cable modem users with a max
> 30KB/s upload ok?
> - What VoIP soft phone works best with Asterisk?
> 
> Also:
> - What kind of uptime are people experiencing?
> - How much system load will be needed for 4 concurrent VoIP conversations?
> - What kind of gotcha's have people had that would be good for a newbie
to
> know?
> 
> 
> Thank you in advance,
> 
> Matt
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
>

> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com 
> [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of 
> Christopher Raper
> Sent: Thursday, 8 January 2004 10:06
> To: asterisk-users@lists.digium.com
> Subject: RE: [Asterisk-Users] Newbie Question-Looking for Feedback
> 
> Greetings all. I am new to the Asterisk world! Found it very 
> impressive so far!
Good.  And you are in Australia too!  For all the seppos who don't know
about Australia, it is just like Iowa except we have pet kangaroos instead
of pitchforks :-)
> In relation to the below.
> 
> I have worked with Alcatel PBX's for the last 3 years. 
> Alcatel OxE supports SIP and H323 as well.
Asterisk does both too, as well as IAX which works very well through NAT.
> As far as SIP goes I have also found the Xlite to be good for 
> soft phones. I am using one now.
> Check out www.xten.com Xlite is free and easy to use. I also 
> have been given a Pingtel SIP to play with.
> http://www.pingtel.com/ 
Many on the list use Xten, and I've heard of Pingtel, I'm not sure how
well
it goes with Asterisk,
many use Grandstream, Snom, or Ci$co hardphones, and DIAX softphone.
> As far as H323 terminals go I have 
> not played with all that many, however the simple Microsoft 
> netmeeting works for testing purpose anyway.
http://openh323.org/ has many H323 apps for linux
> Now a question to all you experts out there, and this may 
> seem VERY stupid, but I have configured the sip phone and 
> have it logged in and can dial 500 to get to the sample 
> messages etc. However i cannot work out how to give the sip 
> termainal a number that can be dialled. I would assume that 
> it needs to be in the dialplan, so I have added it in via the 
> extensions.conf file, however I am sure that I have stuffed 
> the config somewhere. Can someone please point me in the 
> right direction. Would be much appreciated. Also, do i need 
> hardware to make a SIP to SIP call... eg. Compressors etc.
You also need to define the extension in /etc/asterisk/sip.conf

E.g.

[woody]
type=friend
insecure=yes
username=woody
secret=bogus
host=dynamic
defaultip=192.168.2.76

And in extensions.conf:

exten => 1976,1,Dial(SIP/woody,15,tr)

You might want to look at the wiki http://www.voip-info.org/wiki-Asterisk
which is a good place to find out how to do stuff with Asterisk.

cheers,
Woody