Hello, this is a know problem when you are writing the voicemails over a nfs link. you have to start asterisk with the -t option to write voicemail records to the local /tmp and copy it to the final destination after it is finished.
as far as i remember the first 10 seconds are ok and then the speed up started but with the -t option it was completly solved. best regards stefan Am 09.02.12 18:16, schrieb Ruben Rögels: > Hi Dan, > > my wild speculation: It's some kind of timing/synchronisation problem. > Do you use jitter buffer an/or echo cancelation? > > Best regards, > Ruben > > -----Ursprüngliche Nachricht----- > Von: [email protected] > [mailto:[email protected]] Im Auftrag von Dan Ritter > Gesendet: Donnerstag, 9. Februar 2012 17:33 > An: Asterisk Users Mailing List - Non-Commercial Discussion > Betreff: [asterisk-users] Garbled voicemail > > Our Asterisk system (1.8.8.1-1digium1~squeeze) has been very > > stable and generally doing a good job -- except that one day, > > voicemail recordings started being garbled. > > > > It only manifests when the VM comes from our telco gateway > > service -- OnSIP/Junction -- and not from internal phones or > > from an Asterisk box I have at home. > > > > We have voicemail set to record to WAV, and real files are > > being generated -- but it sounds incredibly sped up, faster than > > chipmunks. Completely unintelligible, even if you pull it into > > an audio editor and slow down playback. > > > > It is not perfectly consistent, but it happens in about 85% of > > voicemail recordings left from the outside world through OnSIP. > > > > We've had several years of trouble-free voicemail before this. > > > > Anyone seen anything similar? Advice? Wild speculation? > > > > -dsr- > > > > -- > _____________________________________________________________________ > -- 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 -- Für weitere Fragen stehen wir gerne unter [email protected] oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Teamleiter VOIP // [email protected] // Tel 059944-2440// ------------------------------------------------- SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // ------------------------------------------------- -- _____________________________________________________________________ -- 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
