We have inherited an installation with Ast 1.4 and Aastra phones. The client complains that sometimes the call audio turns tinny and robotic...I heard it and it sounds wierd. Has anyone else experienced this? Cause? Solutions? Thanks, MD -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100212/025f7b72/attachment.htm
What codecs are you using? Are the calls internal(local network) only? Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ----- "Michelle Dupuis" <support at ocg.ca> wrote:>We have inherited an installation with Ast 1.4 and Aastra phones. The client complains that sometimes the call audio turns tinny and robotic...I heard it and it sounds wierd. Has anyone else experienced this? Cause? Solutions? Thanks, MD> -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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/20100212/c8fa2b27/attachment-0001.htm
We experienced this a couple of months ago. It went away when we upgraded the phones to the latest firmware. Another symptom: temporarily putting the caller on hold cures the problem sometimes. Ron Op 12-02-10 22:34, Tim Nelson schreef:> What codecs are you using? Are the calls internal(local network) only? > > Tim Nelson > Systems/Network Support > Rockbochs Inc. > (218)727-4332 x105 > > ----- "Michelle Dupuis" <support at ocg.ca> wrote: > > > We have inherited an installation with Ast 1.4 and Aastra phones. The > client complains that sometimes the call audio turns tinny and > robotic...I heard it and it sounds wierd. > Has anyone else experienced this? Cause? Solutions? > Thanks, > MD > > > -- > _____________________________________________________________________ -- > Bandwidth and Colocation Provided by http://www.api-digital.com -- > asterisk-users mailing list To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- NeoNova BV innovatieve internetoplossingen http://www.neonova.nl Science Park 140 1098 XG Amsterdam info: 020-5611300 servicedesk: 020-5611302 fax: 020-5611301 KvK Amsterdam 34151241 Op dit bericht is de volgende disclaimer van toepassing: http://www.neonova.nl/maildisclaimer
There is a statistics area and you can select sip or voip calls to see calls. It shows packet loss, jitter, latency, out of sequence packets, etc. It can even play them back, so you can check where the loss is and play back the call to see if the noise is in the same spot. Here is some info from the wireshark website: http://wiki.wireshark.org/VoIP_calls -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, February 12, 2010 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Robotic sound sometimes On Fri, 12 Feb 2010, Peder wrote:> Since it is sporadic, my guess would be network latency / packet loss > /jitter to ITSP. You may have lots of capacity and they may claim to > have lots of capacity, but what about the links between you and them. > Who knows when/if there is loss and latency and jitter there. Setup > wireshark to grab some calls and when someone complains about it, look > at the stats for that call. It will tell you if there is loss or > latency or jitter.Do you know of a link that explains how to detect latency or jitter? My wireshark skills are pretty non-existent. Does it have a wizzy filter that will tell you or do you need to check the timestamps of the RTP packets? Any chance you could write up your technique? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
>> We experienced this a couple of months ago. It went away when weupgraded the phones to the latest firmware.>> Another symptom: temporarily putting the caller on hold cures theproblem sometimes. We have Snom 320 phones and had similar problems happening in the call centre intermittently. We resolved by running a weekly script clearing the phones memory. UPGRADES did not work for us. Hope this helps some one. Rudi Oosthuizen