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" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Hardware dimensioning
issues
To: <[EMAIL PROTECTED]>, "'Asterisk Users
Mailing List -
Non-Commercial Discussion'"
<[email protected]>
Message-ID:
<[EMAIL PROTECTED]>
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
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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 <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Overhead Paging
Systems...
To: Asterisk Users Mailing List - Non-Commercial
Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
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" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] ipvolution t1 cards
To: <[email protected]>
Message-ID: <[EMAIL PROTECTED]>
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.
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Message: 3
Date: Thu, 1 Sep 2005 14:44:01 -0400
From: Matt <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] sip jitter buffer in
1.2?
To: Asterisk Users Mailing List - Non-Commercial
Discussion
<[email protected]>
Message-ID:
<[EMAIL PROTECTED]>
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 <[EMAIL PROTECTED]>
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." <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] How to speed-up
INCOMING-RINGING-ENDED
detection on X101P/zapata?
To: "Asterisk Users Mailing List - Non-Commercial
Discussion"
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
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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