<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman, new york, times, serif;font-size:12pt">
Quite surprising, yes! :-)<br>I am from north east Italy, now I live in Verona (Romeo and Juliet's city :).<br>I cannot do it connecting amp to the PBX. I have quite a long distance to cover and a network is already there.<br>My "phones" are have a quite smart processor so we may probably run the ices2 that you are suggesting or something similar.<br>I will check the links that you sent me.<br><br>Thanks,<br>Stefano<br><br><br>>Stefano Totaro,
<br>
><br>>Off topic. I just noticed your name and was a little surprised!? ;-)
<br>>Are you in Italy / Sicily?
<br>
><br>>Anyways, you can achieve overhead paging through a sound card hooked to
<br>>an Amp and speakers from your PBX. I have yet to do it but have read
<br>>about it. I think this may be the better solution for you unless you
<br>>are set on doing it over IP. <br>
><br>>Check here for several options
<br><a rel="nofollow" target="_blank" href="http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom">>http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom</a> <br>
><br>>I am pretty sure if you use a ring group or meetme, there is no way
<br>>around each phone having it's own stream. <br>
><br>>Interestingly, 3Com systems do conferencing and paging through multicast
<br>>which is a nice idea but in practice can be a real pain to configure
<br>>network components to work properly (especially if you do not control
<br>>the network or you are trying to implement paging between remote
<br>>offices). I have spent hours on this exact problem in the past.
<br>
><br>>If it were me, I would probably not want all that traffic on the PBX
<br>>unless that is all that it will be doing or if you go the sound card
<br>>route. I would use ices2 and let everyone stream from a different
<br>>server than the PBX.
<br>
><br>>Since you are using phones, I do not know that ices2 would work for you,
<br>>something must initiate the call. I would probably have a second
<br>>Asterisk box to just handle the paging, setup an extension the dialplan
<br>>of the main PBX to dial the paging machine via SIP (and possibly include
<br>>Authenticate) that would drop the call into something like this:
<br><a rel="nofollow" target="_blank" href="http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page">>http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page</a> <br>
><br>>Thanks,
<br>>Steve Totaro
<br>
><br>
><br>
><br>>>stefano.totaro at transport.alstom.com wrote:
<br>>>
<br>>> Hello Steve,
<br>>> thanks for your anwer.
<br>>> Yes, you are right we want to do VoIP telephone system capable also of
<br>>> "public address" (overhead paging) service.
<br>>> So synchronization is a key issue if we want to avoid unpleasant effects.
<br>>> We are designing our phones and they will have also onboard amplifiers.
<br>>> What I am trying to understand is whether we may use the phone system
<br>>> also for this service or if it is better
<br>>> to go for a specific streaming technology (Ices2 is a good suggestion
<br>>> thanks).
<br>>>
<br>>> What happen if I put all the phones in a ring? Do they join the same
<br>>> multicast stream or a single stream for
<br>>> each phone will be created?
<br>>>
<br>>> Thanks again.
<br>>> Stefano
<br>>>
<br>>>
<br>>> Inactive hide details for Steve Totaro <stevetotaro at hotmail.com>Steve
<br>>> Totaro <stevetotaro at hotmail.com>
<br>>>
<br>>>
<br>>>
<br>>>
<br>>> <br>>> *Steve Totaro <stevetotaro at hotmail.com>*
<br>>> Inviato da:
<br>>> asterisk-users-bounces at lists.digium.com
<br>>>
<br>>>
<br>>> Phone:
<br>>> 01/03/2007 18.33
<br>>> Per favore, rispondere a Asterisk Users
<br>>> Mailing List - Non-Commercial Discussion
<br>>>
<br>>>
<br>>>
<br>>> Per: Asterisk Users Mailing List - Non-Commercial Discussion
<br>>> <asterisk-users at lists.digium.com>
<br>>> Cc: (ccr: Stefano TOTARO/ITVRN01/Transport/ALSTOM)
<br>>>Oggetto: Re: [asterisk-users] Multiple simultaneous calls
<br>>>
<br>>><br>>><br>>>stefano.totaro at transport.alstom.com wrote:
<br>>>>
<br>>>> Hi Guys,
<br>>>> I am a novice of Asterisk and I need some experts help to understand
<br>>>> what I can get out of it.
<br>>>> I need to make multiple calls (let say 50) at once to autoanswering
<br>>>> softphones on a LAN and send all of them the same message that they
<br>>>> will repeat with loudspeakers in the same environment.
<br>>>> I am a little concerned about synchronization of the phones and
<br>>>> moreover it is not much clear to me if I have to open 50 connections
<br>>>> and send 50 times the same packets or if can use in some way the
<br>>>> multicast.
<br>>>> Is there anybody that may give me some idea.
<br>>>> Thanks in advance,
<br>>>> Stefano
<br>>>>
<br>>> I suppose you could do that although, I am unclear on the auto-answering
<br>>> softphone and the loudspeaker thing. Is this just for overhead paging
<br>>> or something?
<br>>>
<br>>> You could put all the phones in a ring group with ringall and use the
<br>>> computer's sound card to connect to an amplified speaker setup.
<br>>>
<br>>> You could also look at ices2 to stream audio or some other streaming
<br>>> technology.
<br>>>
<br>>> Thanks,
<br>>> Steve
<br>>
<br><br>
</div><br>
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