[Asterisk-Users] RE: Hardware dimensioning issues To: <juanmoyano@southecon.com.ar>

Waldo Rubinstein waldo at trianet.net
Thu Sep 1 19:26:04 MST 2005 

Hi Martin,

I read William's and your email and I don't understand your answer.

If I understand Juan's concern, it is the overall ability of the  
server to deliver good quality VoIP services. Both of your  
suggestions to save recorded calls to a database are irrelevant to  
Juan's concern.

If I am wrong, please accept my apologies. However, they way I see  
it, Asterisk still needs to record the file somehow to the file  
system. Whether you run a separate process to move the file from file  
system to a database is a different story. That will only alleviate  
the process of querying for recordings and listening to them.  
However, the direct load on the Asterisk machine will remain, at the  
very least, the same.

The actual question is whether or not he can do what he needs on a  
single 2850 (or any other recommended hardware) or would he need a  
farm of 2850s to spread the load across? If he will need a farm of  
2850s, then Juan's concern should then be focused on how will he be  
able to create conferences across multiple servers. Maybe its  
trivial... I don't know.

Hope my comments help.

Waldo

On Sep 1, 2005, at 9:15 PM, M O wrote:

> Juan,
>
>
> I am running a Calling Card application on a
> Dell PowerEdge 2850 with Asterisk 1.0.7.
>
> Recording conversations I have seen on my server
> causes the processors to burn more than necessary
> so I would recommend what William from Signate
> recommended:
>
> "  Consider saving recorded calls in a database on a
> separate server. It will be simpler to build a
> retrieval interface that does not conflict with
> PBX functions. "
>
> Martin
>
>
> Message: 14
> Date: Thu, 1 Sep 2005 12:39:25 -0700
> From: "William Boehlke" <william.boehlke at signate.com>
> Subject: RE: [Asterisk-Users] Hardware dimensioning
> issues
> To: <juanmoyano at southecon.com.ar>,    "'Asterisk Users
> Mailing List -
>     Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <20050901153927.GA54845 at mail26d.sbc-webhosting.com>
> Content-Type: text/plain;    charset="windows-1250"
>
>
> That's a very ambitious first system.
>
> You may have trouble between the 1850 and the TDM400P.
> The 2850 should be workable.
>
> Consider saving recorded calls in a database on a
> separate server. It will be simpler to build a
> retrieval interface that does not conflict with
> PBX functions.
>
> William Boehlke
> Signate
>
> --- asterisk-users-request at lists.digium.com wrote:
>
>
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>> than "Re: Contents of Asterisk-Users digest..."
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>> Today's Topics:
>>
>>    1. Re: Overhead Paging Systems... (Paul)
>>    2. ipvolution t1 cards (Trey Scarborough)
>>    3. Re: sip jitter buffer in 1.2? (Matt)
>>    4. How to speed-up INCOMING-RINGING-ENDED
>> detection on
>>       X101P/zapata? (Goran Dj.)
>>    5. Re: ztcfg problem (Tzafrir Cohen)
>>    6. Re: /etc/init.d/asterisk barfing (Tzafrir
>> Cohen)
>>    7. Re: /etc/init.d/asterisk barfing (Tzafrir
>> Cohen)
>>    8. Re: ipvolution t1 cards (Andrew Kohlsmith)
>>    9. Re: AGI nor System working after a dial -
>> Should    it    work?
>>       (Patrick Tracanelli)
>>   10. Hardware dimensioning issues (Juan Luis
>> Moyano)
>>   11. Re: /etc/init.d/asterisk barfing (Rich
>> Adamson)
>>   12. IAX2 how to disable VAD ? (Julien)
>>   13. RE: ipvolution t1 cards (Wiley Siler)
>>   14. RE: Hardware dimensioning issues (William
>> Boehlke)
>>   15. Contact Directory on Polycom IP-501 phones
>> (Jesse Keating)
>>   16. Re: Contact Directory on Polycom IP-501 phones
>> (Jeremy Melanson)
>>   17. Re: Realtime IAX (Dana Olson)
>>   18. RE: Speed Questiosn (Carlos Alperin)
>>   19. Re: Contact Directory on Polycom IP-501 phones
>> (Jesse Keating)
>>   20. Re: One way echo canceling? (Matt Fredrickson)
>>   21. Best costs effective solution... (housi
>> mueller)
>>   22. Re: How to shorten ringing stop detection
>> onX101Pclone?
>>       (Goran Dj.)
>>   23. Automon filenames (Anton Krall)
>>   24. RE: Best costs effective solution... (Anton
>> Krall)
>>
>>
>>
>>
> ----------------------------------------------------------------------
>
>>
>> Message: 1
>> Date: Thu, 01 Sep 2005 14:27:13 -0400
>> From: Paul <digium-list at 9ux.com>
>> Subject: Re: [Asterisk-Users] Overhead Paging
>> Systems...
>> To: Asterisk Users Mailing List - Non-Commercial
>> Discussion
>>     <asterisk-users at lists.digium.com>
>> Message-ID: <43174801.6030906 at 9ux.com>
>> Content-Type: text/plain; charset=windows-1250;
>> format=flowed
>>
>> William Boehlke wrote:
>>
>>
>>> Viking makes everything you might need for paging
>>>
>> and door control.
>>
>>> www.vikingtelecomsolutions.com
>>>
>>> William Boehlke
>>> Signate
>>>
>>>
>>>
>> I have one customer with a nortel meridian pbx and
>> there is viking stuff
>> all over the backboard. I never had to mess with any
>> of it because it
>> all works as intended.
>>
>>
>>
>> ------------------------------
>>
>> Message: 2
>> Date: Thu, 1 Sep 2005 13:27:22 -0500
>> From: "Trey Scarborough" <treys at door.net>
>> Subject: [Asterisk-Users] ipvolution t1 cards
>> To: <asterisk-users at lists.digium.com>
>> Message-ID: <040201c5af22$cbda2ff0$5f00080a at treypc>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> Has any one used the Ipvolution tdm120 cards i am
>> intrested to know how well it works and how well the
>> on board dsp's work.
>> -------------- next part --------------
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>> ------------------------------
>>
>> Message: 3
>> Date: Thu, 1 Sep 2005 14:44:01 -0400
>> From: Matt <mhoppes at gmail.com>
>> Subject: Re: [Asterisk-Users] sip jitter buffer in
>> 1.2?
>> To: Asterisk Users Mailing List - Non-Commercial
>> Discussion
>>     <asterisk-users at lists.digium.com>
>> Message-ID:
>> <c11d025305090111446af0f405 at mail.gmail.com>
>> Content-Type: text/plain; charset=ISO-8859-1
>>
>> I am using it with CVS-HEAD.... but it is currently
>> a patch.  So far
>> the version of the patch I have (which was the first
>> one released)..
>> seems to be working very well.. and definately makes
>> a noticeable
>> improvement.
>>
>> On 9/1/05, Damon Estep <damon at suburbanbroadband.net>
>> wrote:
>>
>>>
>>>
>>>
>>> Did the sip jitter buffer make it into 1.2? anyone
>>>
>> using it?
>>
>>> _______________________________________________
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>>
>> ------------------------------
>>
>> Message: 4
>> Date: Thu, 1 Sep 2005 20:48:10 +0200
>> From: "Goran Dj." <pisac at hotpop.com>
>> Subject: [Asterisk-Users] How to speed-up
>> INCOMING-RINGING-ENDED
>>     detection on    X101P/zapata?
>> To: "Asterisk Users Mailing List - Non-Commercial
>> Discussion"
>>     <asterisk-users at lists.digium.com>
>> Message-ID: <007201c5af25$b500b3a0$0300a8c0 at gogi>
>> Content-Type: text/plain;    charset="iso-8859-2"
>>
>>
>>> Pause betwen incoming rings on my phone line is
>>>
>> 4s, so when x101p
>> clone
>>
>>> (wcfxo driver) do not receive next ring signal
>>>
>> after 4.5 sec, call
>>
>>> should be consider as ended.
>>>
>>> What should I change to set that time (4.5 sec)
>>>
>> for incoming ring end
>>
>>> detection?
>>>
>>
>> I measured, event "-- Hungup 'Zap/1-1'" is shown
>> exactly 8 sec after
>> last detected ring (on X101P), and my voip phone
>> continues to ringing
>> during that time (that's bad). I want to cut that
>> time to 4.5 sec. How
>> to do that?
>>
>> I tried to change in zapata.h some lines:
>> #define ZT_DEFAULT_RINGTIME 500
>> #define ZT_LOOPCODE_TIME 3000
>> #define ZT_RINGOFFTIME 2000
>> but with no effects. "Hungup" is still shown 8 sec
>> after last ring.
>>
>>
>>
>>
>>
>>
> === message truncated ===
>
>
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