Hello Everyone, We are currently experiencing some higher load on our servers, and since signaling comes into our servers on G729, we would like to implement G729 pass-through. A few questions arise, do we need to convert all the recording to the codec, and what about voicemail? We are also using A2Billing (hope I am not violating any thread rules), and would like to convert all that recording to G729 as well. Any help is greatly appreciated. Kind Regards, Nick from Toronto.
Anyone? :) N. On 8/13/13, Nick Khamis <symack at gmail.com> wrote:> Hello Everyone, > > We are currently experiencing some higher load on our servers, and > since signaling comes into our servers on G729, we would like to > implement G729 pass-through. A few questions arise, do we need to > convert all the recording to the codec, and what about voicemail? > > We are also using A2Billing (hope I am not violating any thread > rules), and would like to convert all that recording to G729 as well. > > Any help is greatly appreciated. > > Kind Regards, > > Nick from Toronto. >
Sent from my Verizon Wireless 4G LTE DROID Eric Wieling <EWieling at nyigc.com> wrote:>If you want to do pass-thu you need to make sure Asterisk NEVER EVER needs to transcode. This means all calls must use only g729, sound files must be in g729 format and no early audio, inband ringing or anything else which might cause Asterisk to require a temp transcoding path. > >In my experience it never works right. The most you should expect to be able to do is reduce the need for transcoding by doing the above steps. > >-----Original Message----- >From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nick Khamis >Sent: Wednesday, August 14, 2013 10:20 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [asterisk-users] G729 Passthrough How To > >Anyone? :) > >N. > >On 8/13/13, Nick Khamis <symack at gmail.com> wrote: >> Hello Everyone, >> >> We are currently experiencing some higher load on our servers, and >> since signaling comes into our servers on G729, we would like to >> implement G729 pass-through. A few questions arise, do we need to >> convert all the recording to the codec, and what about voicemail? >> >> We are also using A2Billing (hope I am not violating any thread >> rules), and would like to convert all that recording to G729 as well. >> >> Any help is greatly appreciated. >> >> Kind Regards, >> >> Nick from Toronto. >> > >-- >_____________________________________________________________________ >-- 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/hello > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-- >_____________________________________________________________________ >-- 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/hello > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130828/37d26cf2/attachment.htm>
Sent from my Verizon Wireless 4G LTE DROID Eric Wieling <EWieling at nyigc.com> wrote:>If you want to do pass-thu you need to make sure Asterisk NEVER EVER needs to transcode. This means all calls must use only g729, sound files must be in g729 format and no early audio, inband ringing or anything else which might cause Asterisk to require a temp transcoding path. > >In my experience it never works right. The most you should expect to be able to do is reduce the need for transcoding by doing the above steps. > >-----Original Message----- >From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nick Khamis >Sent: Wednesday, August 14, 2013 10:20 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [asterisk-users] G729 Passthrough How To > >Anyone? :) > >N. > >On 8/13/13, Nick Khamis <symack at gmail.com> wrote: >> Hello Everyone, >> >> We are currently experiencing some higher load on our servers, and >> since signaling comes into our servers on G729, we would like to >> implement G729 pass-through. A few questions arise, do we need to >> convert all the recording to the codec, and what about voicemail? >> >> We are also using A2Billing (hope I am not violating any thread >> rules), and would like to convert all that recording to G729 as well. >> >> Any help is greatly appreciated. >> >> Kind Regards, >> >> Nick from Toronto. >> > >-- >_____________________________________________________________________ >-- 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/hello > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-- >_____________________________________________________________________ >-- 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/hello > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130828/b0bfdc08/attachment.htm>