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[bk1379 at yahoo.com]

HiI am evaluating some voice broadcasting solutions based on Asterisks for more than 1000 simultaneous calls.Connection to Asterisk all are based on SIP and SIP Trunks so no DAHDI hardware is required.According to some recommendations like http://osdial.org/howto/ "Internal timing is very critical with Asterisk when it is under load"and we must use DAHDI hardware or "USB Voice Synch Tool" http://www.sangoma.com/accessories/specialty-tools/But according to my understanding of wiki https://wiki.asterisk.org/wiki/display/AST/Timing+InterfacesIt seems it is not necessary now.Please tell me your opinions.Regards

[sgriepentrog at digium...]

If you were running on Asterisk 1.4, a Zaptel or Dahdi timing source (including the Sangoma USB device) was necessary to avoid sometimes unreliable timing from the "dummy" interface.


For modern releases (1.6, 1.8, 11, 12, etc) this isn't necessary for most systems.  However, you may have better results with such a large number of calls by using a hardware timing source.  The difference will vary between different systems and loads -- I recommend testing it on your own platform.  Note that changing to a different model with a different motherboard or even just a different chipset can result in a difference in timing accuracy.


 -- so your best option is to try it both ways under load to see if you see a benefit, and re-test should you change the platform, such as using a different motherboard.





On Wed, Jul 30, 2014 at 4:08 AM, babak <bk1379@yahoo.com (bk1379@yahoo.com)> wrote:

[asterisk.org at sedwar...]

On Wed, 30 Jul 2014, babak wrote:

[bk1379 at yahoo.com]

Thank you for replys
1300 calls include playback voices ?
which timing module you are using:
res_timing_timerfd.so orres_timing_kqueue.so or res_timing_dahdi.sores_timing_pthread.so
Regards




On Thursday, July 31, 2014 12:16 AM, Steve Edwards <asterisk.org@sedwards.com> wrote:
On Wed, 30 Jul 2014, babak wrote:> According to some recommendations like http://osdial.org/howto/> "Internal timing is very critical with Asterisk when it is under load" > and we must use DAHDI hardware or "USB Voice Synch Tool" > http://www.sangoma.com/accessories/specialty-tools/But according to my > understanding of wiki > https://wiki.asterisk.org/wiki/display/AST/Timing+InterfacesIt seems it > is not necessary now. Please tell me your opinions.
I'm running Asterisk 11.11.0 on an HP ProLiant DL320e Gen8 E3-1240v2.1,300 calls with no audio issues.-- Thanks in advance,-------------------------------------------------------------------------Steve Edwards sedwards@sedwards.com (sedwards@sedwards.com) Voice: +1-760-468-3867 PSTNewline Fax: +1-760-731-3000-- _____________________________________________________________________-- Bandwidth and Colocation Provided by http://www.api-digital.com--New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/helloasterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk.org at sedwar...]

Please don't top post.

Please keep the thread only on the list.

[mjordan at digium.com]

On Wed, Jul 30, 2014 at 8:33 PM, Steve Edwards <asterisk.org@sedwards.com (asterisk.org@sedwards.com)> wrote:


<snip>

 

[asterisk at lists.mino...]

On 30/7/14 10:08 am, babak wrote:

[dtaylor at vocalabs.com]

On 07/31/2014 09:51 AM, Chris Bagnall wrote:

[bk1379 at yahoo.com]