I currently am running on a Intel(R) Xeon(R) CPU E5-2670 v2 @ 2.50GHz
Codec im using is codec_g729-ast18-icc-glibc-x86_64-core2.so Kind Regards Andrew Colin Converged Data (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Direct: +27 (0)10 591 4607 Mobile: +27 (0)82 310 3007 Switchboard: +27 (0)10 591 4600 Email: and...@convergedgroup.net Web: http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof. From: Mitul Limbani [mailto:mi...@enterux.in] Sent: Friday, November 21, 2014 1:04 PM To: Andrew Colin Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] One way audio internal Then something to do with your codec selection priority. On 21-Nov-2014 4:26 PM, "Andrew Colin" <and...@convergedgroup.net> wrote: I am using the free g729 Kind Regards Andrew Colin Converged Data (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Direct: +27 (0)10 591 4607 <tel:%2B27%20%280%2910%20591%204607> Mobile: +27 (0)82 310 3007 <tel:%2B27%20%280%2982%20310%203007> Switchboard: +27 (0)10 591 4600 <tel:%2B27%20%280%2910%20591%204600> Email: and...@convergedgroup.net Web: http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof. From: Mitul Limbani [mailto:mi...@enterux.in] Sent: Friday, November 21, 2014 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Andrew Colin Subject: Re: [asterisk-users] One way audio internal You probably do not have enough g729 channels license. On 21-Nov-2014 4:17 PM, "A J Stiles" <asterisk_l...@earthshod.co.uk> wrote: On Friday 21 Nov 2014, Andrew Colin wrote: > Hi All > > We have a strange issue with our hosted asterisk server running on Debian > Internal calls btween extensions using g729 give one way audio > As soon as we change the codec to ALAW the issues goes away. > > Any ideas how to fix this? > > Outbound calls via a trunk work fine with g729 Unless you have serious bandwidth issues, just forget about g.729 and change to a-law throughout. A-law is what the PSTN (in civilised countries) uses anyway, so you won't need to transcode (which chews up processor resources and risks compromising quality) for calls to and from the outside world. If you really need to use g.729 and are outside the USA (therefore, beyond the reach of software patents), there is a free version that you can use -- and this one, better than Digium's offering, comes with the Source Code so you can be sure it isn't doing anything nasty behind the scenes. But to be honest, you probably are better off just sticking with a-law. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _____________________________________________________________________ -- 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