M O
2005-Sep-01 18:15 UTC
[Asterisk-Users] RE: Hardware dimensioning issues To: <juanmoyano@southecon.com.ar>
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@signate.com> Subject: RE: [Asterisk-Users] Hardware dimensioning issues To: <juanmoyano@southecon.com.ar>, "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <20050901153927.GA54845@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@lists.digium.com wrote:> Send Asterisk-Users mailing list submissions to > asterisk-users@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@lists.digium.com > > You can reach the person managing the list at > asterisk-users-owner@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@9ux.com> > Subject: Re: [Asterisk-Users] Overhead Paging > Systems... > To: Asterisk Users Mailing List - Non-Commercial > Discussion > <asterisk-users@lists.digium.com> > Message-ID: <43174801.6030906@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@door.net> > Subject: [Asterisk-Users] ipvolution t1 cards > To: <asterisk-users@lists.digium.com> > Message-ID: <040201c5af22$cbda2ff0$5f00080a@treypc> > Content-Type: text/plain; charset="iso-8859-1" > > Has any one used the Ipvolution tdm120 cards i am > intrested to know how well it works and how well the > on board dsp's work. > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: >http://lists.digium.com/pipermail/asterisk-users/attachments/20050901/208c5541/attachment-0001.htm> > ------------------------------ > > Message: 3 > Date: Thu, 1 Sep 2005 14:44:01 -0400 > From: Matt <mhoppes@gmail.com> > Subject: Re: [Asterisk-Users] sip jitter buffer in > 1.2? > To: Asterisk Users Mailing List - Non-Commercial > Discussion > <asterisk-users@lists.digium.com> > Message-ID: > <c11d025305090111446af0f405@mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I am using it with CVS-HEAD.... but it is currently > a patch. So far > the version of the patch I have (which was the first > one released).. > seems to be working very well.. and definately makes > a noticeable > improvement. > > On 9/1/05, Damon Estep <damon@suburbanbroadband.net> > wrote: > > > > > > > > Did the sip jitter buffer make it into 1.2? anyone > using it? > > _______________________________________________ > > --Bandwidth and Colocation sponsored by > Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > >http://lists.digium.com/mailman/listinfo/asterisk-users> > To UNSUBSCRIBE or update options visit: > > > > >http://lists.digium.com/mailman/listinfo/asterisk-users> > > > > > > ------------------------------ > > Message: 4 > Date: Thu, 1 Sep 2005 20:48:10 +0200 > From: "Goran Dj." <pisac@hotpop.com> > Subject: [Asterisk-Users] How to speed-up > INCOMING-RINGING-ENDED > detection on X101P/zapata? > To: "Asterisk Users Mailing List - Non-Commercial > Discussion" > <asterisk-users@lists.digium.com> > Message-ID: <007201c5af25$b500b3a0$0300a8c0@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
Waldo Rubinstein
2005-Sep-01 19:26 UTC
[Asterisk-Users] RE: Hardware dimensioning issues To: <juanmoyano@southecon.com.ar>
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@signate.com> > Subject: RE: [Asterisk-Users] Hardware dimensioning > issues > To: <juanmoyano@southecon.com.ar>, "'Asterisk Users > Mailing List - > Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: > <20050901153927.GA54845@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@lists.digium.com wrote: > > >> Send Asterisk-Users mailing list submissions to >> asterisk-users@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@lists.digium.com >> >> You can reach the person managing the list at >> asterisk-users-owner@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@9ux.com> >> Subject: Re: [Asterisk-Users] Overhead Paging >> Systems... >> To: Asterisk Users Mailing List - Non-Commercial >> Discussion >> <asterisk-users@lists.digium.com> >> Message-ID: <43174801.6030906@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@door.net> >> Subject: [Asterisk-Users] ipvolution t1 cards >> To: <asterisk-users@lists.digium.com> >> Message-ID: <040201c5af22$cbda2ff0$5f00080a@treypc> >> Content-Type: text/plain; charset="iso-8859-1" >> >> Has any one used the Ipvolution tdm120 cards i am >> intrested to know how well it works and how well the >> on board dsp's work. >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: >> >> > http://lists.digium.com/pipermail/asterisk-users/attachments/ > 20050901/208c5541/attachment-0001.htm > >> >> ------------------------------ >> >> Message: 3 >> Date: Thu, 1 Sep 2005 14:44:01 -0400 >> From: Matt <mhoppes@gmail.com> >> Subject: Re: [Asterisk-Users] sip jitter buffer in >> 1.2? >> To: Asterisk Users Mailing List - Non-Commercial >> Discussion >> <asterisk-users@lists.digium.com> >> Message-ID: >> <c11d025305090111446af0f405@mail.gmail.com> >> Content-Type: text/plain; charset=ISO-8859-1 >> >> I am using it with CVS-HEAD.... but it is currently >> a patch. So far >> the version of the patch I have (which was the first >> one released).. >> seems to be working very well.. and definately makes >> a noticeable >> improvement. >> >> On 9/1/05, Damon Estep <damon@suburbanbroadband.net> >> wrote: >> >>> >>> >>> >>> Did the sip jitter buffer make it into 1.2? anyone >>> >> using it? >> >>> _______________________________________________ >>> --Bandwidth and Colocation sponsored by >>> >> Easynews.com -- >> >>> >>> Asterisk-Users mailing list >>> Asterisk-Users@lists.digium.com >>> >>> >> >> > http://lists.digium.com/mailman/listinfo/asterisk-users > >>> To UNSUBSCRIBE or update options visit: >>> >>> >>> >> >> > http://lists.digium.com/mailman/listinfo/asterisk-users > >>> >>> >>> >> >> >> ------------------------------ >> >> Message: 4 >> Date: Thu, 1 Sep 2005 20:48:10 +0200 >> From: "Goran Dj." <pisac@hotpop.com> >> Subject: [Asterisk-Users] How to speed-up >> INCOMING-RINGING-ENDED >> detection on X101P/zapata? >> To: "Asterisk Users Mailing List - Non-Commercial >> Discussion" >> <asterisk-users@lists.digium.com> >> Message-ID: <007201c5af25$b500b3a0$0300a8c0@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 > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@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 >