[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #1918 - 9 msgs

Mon Nov 17 16:16:09 MST 2003

An example for Radius is calling cards.. I can use * for this kind of
service... With platforms that use Radius Server.

-----Mensaje original-----
De: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] En nombre de
asterisk-users-request at lists.digium.com
Enviado el: Lunes, 17 de Noviembre de 2003 07:16 p.m.
Para: asterisk-users at lists.digium.com
Asunto: Asterisk-Users digest, Vol 1 #1918 - 9 msgs

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Today's Topics:

   1. RE: wireless (mick at netexpress.com.au)
   2. Re: Radius on * (Jeremy McNamara)
   3. Re: IAX2 connectivity problem (qualify=yes) (Andrew Thompson)
   4. Re: VOIP phonesets vs. cheap Analog touch-tone
       sets with Asterisk (Howard White)
   5. Re: SIP calls no longer work (Andrew Thompson)
   6. 3Com NBX phones (Andrew Nelson)
   7. asterisk and Codec G-723 (Javier Rios)
   8. Re: DTMF (Scott England)

--__--__--

Message: 1
From: <mick at netexpress.com.au>
To: <asterisk-users at lists.digium.com>
Subject: RE: [Asterisk-Users] wireless
Date: Tue, 18 Nov 2003 08:03:06 +1030
Reply-To: asterisk-users at lists.digium.com

Don't sound bad do they

Except their ship date is mid January 2004

Regards Mick 

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Iain
Stevenson
Sent: Monday, 17 November 2003 10:34 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] wireless

AFAIK the 7920 needs CallManager to work - if you haven't got that
you'll 
have to wait for Cisco to make a general purpose version - or maybe buy
a 
Pulver phone <http://www.pulverinnovations.com/> - assuming that works
with 
*

  Iain

--On Monday, November 17, 2003 6:31 am -0500 Jeremy McNamara 
<jj at nufone.net> wrote:

> mick at netexpress.com.au wrote:
>
>> Has anyone got a mobile wireless phone working with * yet ????
>>
>> Is it possible to use the Cisco 7920 with skinny ????
>>
>>
>
> Not sure, send me one and I'll test it for you.
>
>
> Jeremy McNamara
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

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

--__--__--

Message: 2
Date: Mon, 17 Nov 2003 16:33:10 -0500
From: Jeremy McNamara <jj at nufone.net>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Radius on *
Reply-To: asterisk-users at lists.digium.com

Sebastian Nocetti wrote:

>Does Asterisk support Radius accounting?....
>  
>

No and there is absolutely no need for it to.   RADIUS is not anything 
that should have ever been deployed in a VoIP environment.  

There are many methods to talk directly to a database, why add another 
layer of complexity and point of failure?

Jeremy McNamara

--__--__--

Message: 3
From: "Andrew Thompson" <asteriskuser at aktzero.com>
To: <asterisk-users at lists.digium.com>
Subject: Re: [Asterisk-Users] IAX2 connectivity problem (qualify=yes)
Date: Mon, 17 Nov 2003 16:54:06 -0500
Reply-To: asterisk-users at lists.digium.com

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--__--__--

Message: 4
Subject: Re: [Asterisk-Users] VOIP phonesets vs. cheap Analog touch-tone
	sets with Asterisk
From: Howard White <howwhi at vcch.com>
To: asterisk-users at lists.digium.com
Date: 17 Nov 2003 15:56:53 -0600
Reply-To: asterisk-users at lists.digium.com

<bottom response = on>

On Mon, 2003-11-17 at 08:34, Steve Murphy wrote:
> Hello--
> 
> I've been asked an interesting question, and I'm too ignorant to
answer
> it authoritatively (yet). Can anyone help me?
> 
> Question: If I'm going to implement a somewhat small (10-80) phone
> system, and I have a choice of using VOIP phoneset (like SNOM or
> Grandstream or Cisco, etc), vs. cheap analog touch-tone phones,
exactly
> what features will I kiss goodbye if I use the cheap analogs?
> 
> In other words, what features will a (more expensive) VOIP phoneset
> provide, that the analog won't?
> 
> I know already that asterisk will give me these features with just
plain
> analog phones (&zaptel cards, of course): Voice mail, park & retrieve
&
> MOH, transfer, agents, and a few others. And, if you get an analog
with
> a CID built in, you could have that, too? (Haven't tried that yet).
> 
> What is the justification for VOIP? just total cost reductions (if the
> phone is cheap enough?)? Or are there some nicenesses that only VOIPs
> can supply?
> 
> murf

Let's approach this question from economics first (tacitly assuming that
the technical issues wash).  This "debate" is central to what most of us
(consultants) are facing just now.  The short answer is that the mission
may be accomplished either way.

Start out by lining up the quantities, types and costs of the equipment
required.  

With an analog phone deployment, anything beyond four desksets is best
accomplished with T1 card(s) in the Asterisk PC and channel banks to
connect the phones (and probably the PSTN).  Whereas the phones could be
as little as $10 each, you have to budget the price of the channel
bank(s).  If the customer is uncomfortable with eBay, used hardware,
channel banks could be north of $3000 each.

With a net-phone deployment, channel banks are out but more and better
ethernet switches _may_ be chosen (specifically taking advantage of
QoS/ToS) or separate voice-IP networks installed alongside data-IP
networks.  For ten desksets, probably not but for eighty???

I have to admit to being quite agnostic to leaning toward analog,
myself.  If Jeff Pulver's demand (at VON this September) that SIP phone
costs be brought down to the $20 per deskset level is met, the economics
change quickly.

Howard White
president - VCCH, Inc.

--__--__--

Message: 5
From: "Andrew Thompson" <asteriskuser at aktzero.com>
To: <asterisk-users at lists.digium.com>
Subject: Re: [Asterisk-Users] SIP calls no longer work
Date: Mon, 17 Nov 2003 16:57:04 -0500
Reply-To: asterisk-users at lists.digium.com

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--__--__--

Message: 6
From: Andrew Nelson <nelsonab at pobox.com>
To: asterisk-users at lists.digium.com
Date: Mon, 17 Nov 2003 14:02:53 -0800
Subject: [Asterisk-Users] 3Com NBX phones
Reply-To: asterisk-users at lists.digium.com

Has there been any luck getting the 3Com NBX series phones to work with 
Asterisk?

Thanks!

-Andrew

--__--__--

Message: 7
From: "Javier Rios" <javierrios at cantv.net>
To: <asterisk-users at lists.digium.com>
Date: Mon, 17 Nov 2003 18:06:13 -0400
Subject: [Asterisk-Users] asterisk and Codec G-723
Reply-To: asterisk-users at lists.digium.com

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Hello

I want that all the information of voice that travels by asterisk works
with codec g-723 

This can be done? 

Javier Rios

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<body lang=3DES link=3Dblue vlink=3Dpurple =
style=3D'tab-interval:35.4pt'>

<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Hello<o:p></o:p></span
>=
</font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></spa
n=
></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3DArial><span =
lang=3DEN
style=3D'font-size:10.0pt;font-family:Arial;color:black;mso-ansi-languag
e=
:EN'>I
want that all the information of voice that travels by asterisk works =
with
codec g-723 <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3DArial><span =
lang=3DEN
style=3D'font-size:10.0pt;font-family:Arial;color:black;mso-ansi-languag
e=
:EN'><o:p>&nbsp;</o:p></span></font></p>

This
can be done? <o:p></o:p>

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lang=3DEN-GB
style=3D'font-size:12.0pt;mso-ansi-language:EN-GB;mso-no-proof:yes'>&nbs
p=
;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
lang=3DEN-GB
style=3D'font-size:12.0pt;mso-ansi-language:EN-GB'>Javier =
Rios<o:p></o:p></span></font></p>

</div>

</body>

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Message: 8
Date: Mon, 17 Nov 2003 14:00:23 -0800
From: Scott England <scott at homelan.com>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] DTMF
Reply-To: asterisk-users at lists.digium.com

This is a multi-part message in MIME format.
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Vocal is a complex system and its very Cisco centric (since Cisco funds 
it I'm not suprised) but in all its been good. We went with it for the 
SIP support and the fact that our version is heavily modified to 
intgrate with our custom app. If * had solid SIP support 2 years ago 
things might have been diffrent. :)

Scott

costas wrote:

>I can't resist asking. What do you think of Vocal as compared to *?
Anything Vocal has but missing in *?
>
>---------- Original Message ----------------------------------
>From: Scott England <scott at homelan.com>
>Reply-To: asterisk-users at lists.digium.com
>Date:  Mon, 17 Nov 2003 02:58:55 -0800
>
>  
>
>>I am trying to connect to a vocal server from an asterisk server. A
call 
>>is received via iax2 to my asterisk server. I then initiate a SIP 
>>connection to the vocal server. everything works great except dtmf 
>>doesnt work. A cisco 5300 can connect to this vocal server and do dtmf

>>without a problem. I have my dtmf set to rfc2833 in the general
section 
>>of the sip.conf . I can confirm that the channel is in rfc2833 during 
>>the call via show channel. With SIP debug though I dont see any event 
>>for dtmf. I do see dtmf in IAX though.
>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>    
>>
>
>--
>Costas Menico
>Meezon Software Corp
>201-224-8111
>costas at meezon.com
>
>--
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>  
>

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content="text/html;charset=ISO-8859-1">
 <title></title>
</head>
<body text="#000000" bgcolor="#ffffff">
Vocal is a complex system and its very Cisco centric (since Cisco funds
it I'm not suprised) but in all its been good. We went with it for the
SIP support and the fact that our version is heavily modified to
intgrate with our custom app. If * had solid SIP support 2 years ago
things might have been diffrent. :) 
 
Scott 
 
costas wrote: 
<blockquote type="cite"
 cite="mid200311171546.AA432411198 at mail.meezon.com">
 <pre wrap="">I can't resist asking. What do you think of Vocal as
compared to *? Anything Vocal has but missing in *?

---------- Original Message ----------------------------------
From: Scott England <a class="moz-txt-link-rfc2396E"
href="mailto:scott at homelan.com">&lt;scott at homelan.com&gt;</a>
Reply-To: <a class="moz-txt-link-abbreviated"
href="mailto:asterisk-users at lists.digium.com">asterisk-users at lists.digiu
m.com</a>
Date:  Mon, 17 Nov 2003 02:58:55 -0800

  </pre>
  <blockquote type="cite">
    <pre wrap="">I am trying to connect to a vocal server from an
asterisk server. A call 
is received via iax2 to my asterisk server. I then initiate a SIP 
connection to the vocal server. everything works great except dtmf 
doesnt work. A cisco 5300 can connect to this vocal server and do dtmf 
without a problem. I have my dtmf set to rfc2833 in the general section 
of the sip.conf . I can confirm that the channel is in rfc2833 during 
the call via show channel. With SIP debug though I dont see any event 
for dtmf. I do see dtmf in IAX though.

_______________________________________________
Asterisk-Users mailing list
<a class="moz-txt-link-abbreviated"
href="mailto:Asterisk-Users at lists.digium.com">Asterisk-Users at lists.digiu
m.com</a>
<a class="moz-txt-link-freetext"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://li
sts.digium.com/mailman/listinfo/asterisk-users</a>

    </pre>
  </blockquote>
  <pre wrap=""><!---->
--
Costas Menico
Meezon Software Corp
201-224-8111
<a class="moz-txt-link-abbreviated"
href="mailto:costas at meezon.com">costas at meezon.com</a>

--
_______________________________________________
Asterisk-Users mailing list
<a class="moz-txt-link-abbreviated"
href="mailto:Asterisk-Users at lists.digium.com">Asterisk-Users at lists.digiu
m.com</a>
<a class="moz-txt-link-freetext"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://li
sts.digium.com/mailman/listinfo/asterisk-users</a>
  </pre>
</blockquote>
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