asterisk users - Dec 2006 - Server for 100 concurrent calls

Hi all,

I'm looking at some suggestions from you techies out there.

Let me explain my scenario. Im a reseller to callshops.

I need to take around 100 concurrent calls. Almost all endpoints are sending
G723 codec and my peers take G729.

Can anyone recommend the Server Specs that is ideal for this scenario. Im
planning to lease a server. Calls are purely SIP or IAX2 only.

Thanks in advance.

Dan
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sdcharly@gmail.com a ?crit :
> Hi all,
>  
> I'm looking at some suggestions from you techies out there.
>  
> Let me explain my scenario. Im a reseller to callshops.
>  
> I need to take around 100 concurrent calls. Almost all endpoints are 
> sending G723 codec and my peers take G729.
Since Digium doesn't provide g723 codecs (as far as I'm aware), and 
there's yet no transcoding card for Asterisk (one is supposed to be out 
at some point, but when... god knows), for the moment you should look 
into something else than Asterisk.

Cheers,
Jean-Michel.

Hi Jerry, THANKS A LOT.
I viewed configuration files so many times, but I had to be blind so I
didn't noticed that mistake. I was solving this problem for almost two days
with no success... thanks a lot again. :)

It can sound weird, but I cannot wait for Monday when I go to work... :D

Petosh


----- Original Message ----- 
From: "Jerry" <jerry@voiptower.com>
To: "Petr Kovar" <petoshasterisk@centrum.cz>
Sent: Friday, December 08, 2006 4:44 PM
Subject: Re: [asterisk-users] TDM400 and analog phone - can't dial

> Hi Petosh,
>
>> Hi all,
>> I have a problem with dialing digits from my analog phone connected to
>> TDM400 with one FXS card. I can call the phone from SIP, but when I try
>> [...]
>> I'm newbie to Asterisk so, please, can someone check my
configuration
>> and tell me I have everything alright (I think its ok, I did it the
same
>> way as in asterisk TFOT book) and I can focus to grab theory for DTFM
>> problem through weekend?
>
> It's not DTMF.
>
>> my configuration files:
>>
>> *extensions.conf:*
>>
>> [internal]
>> exten => 101,1,Dial(SIP/petosh,20)
>> exten => 101,2,Playback(my/notavailable)
>> exten => 101,3,Hangup( )
>>
>> exten => 200,1,Dial(Zap/1,20)
>> exten => 200,2,Playback(my/notavailable)
>> exten => 200,3,Hangup( )
> [plus stuff omitted ... important later]
>
>> *zapata.conf:
>> *[trunkgroups]
>>
>> [channels]
>> usercallerid=yes
>> hidecallerid=no
>> callwaiting=no
>> threewaycalling=yes
>> transfer=yes
>> echocancel=yes
>> echotraining=yes
>>
>> ; *** Added by me - note the next line (which is in your original)
>> context=incoming
>> signalling=fxo_ks
>> channel => 1
>
> BANG! and there it is.
>
> Either you are missing a context of "incoming" in your
extensions.conf, or
> you didn't list it.
>
> Whatever the case, Asterisk is trying to dial based on this
"incoming"
> context. You can change it to "internal", but realize how it
works ... it
> is the part of the dialplan where the digits you are dialing are searched
> for. If you just cut and pasted from the FOT, it could be they were using
> an FXO (connected to a phone line), and thus the "incoming"
context was
> for answering the phone.
>
> Hope that makes sense.
>
> Thanks,
>
> J.
>