[Asterisk-Users] RE: Hardware dimensioning issues To:
<juanmoyano@southecon.com.ar>
M O
martinoshield at yahoo.com
Thu Sep 1 18:15:40 MST 2005
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:
> Send Asterisk-Users mailing list submissions to
> asterisk-users at lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web,
> visit
>
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body
> 'help' to
> asterisk-users-request at lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-owner at lists.digium.com
>
> When replying, please edit your Subject line so it
> is more specific
> than "Re: Contents of Asterisk-Users digest..."
>
>
> 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$5f00080a at treypc>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Has any one used the Ipvolution tdm120 cards i am
> intrested to know how well it works and how well the
> on board dsp's work.
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL:
>
http://lists.digium.com/pipermail/asterisk-users/attachments/20050901/208c5541/attachment-0001.htm
>
> ------------------------------
>
> Message: 3
> Date: Thu, 1 Sep 2005 14:44:01 -0400
> From: Matt <mhoppes at gmail.com>
> Subject: Re: [Asterisk-Users] sip jitter buffer in
> 1.2?
> To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <c11d025305090111446af0f405 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> I am using it with CVS-HEAD.... but it is currently
> a patch. So far
> the version of the patch I have (which was the first
> one released)..
> seems to be working very well.. and definately makes
> a noticeable
> improvement.
>
> On 9/1/05, Damon Estep <damon at suburbanbroadband.net>
> wrote:
> >
> >
> >
> > Did the sip jitter buffer make it into 1.2? anyone
> using it?
> > _______________________________________________
> > --Bandwidth and Colocation sponsored by
> Easynews.com --
> >
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
>
> ------------------------------
>
> Message: 4
> Date: Thu, 1 Sep 2005 20:48:10 +0200
> From: "Goran Dj." <pisac at hotpop.com>
> Subject: [Asterisk-Users] How to speed-up
> INCOMING-RINGING-ENDED
> detection on X101P/zapata?
> To: "Asterisk Users Mailing List - Non-Commercial
> Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <007201c5af25$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 ===
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
More information about the asterisk-users
mailing list