On Monday 13 September 2010 06:07:09 Sebastian wrote: > On 09/13/2010 11:34 AM, Olivier wrote: > > In voicemail.conf.sample, you can read this: > > > > format=wav49|gsm|wav > > ; WARNING: > > ; If you change the list of formats that you record voicemail in > > ; when you have mailboxes that contain messages, you _MUST_ absolutely > > ; manually go through those mailboxes and convert/delete/add the > > ; the message files so that they appear to have been stored using > > ; your new format list. If you don't do this, very unpleasant > > ; things may happen to your users while they are retrieving and > > ; manipulating their voicemail. > > ; > > ; In other words: don't change the format list on a production system > > ; unless you are _VERY_ sure that you know what you are doing and are > > ; prepared for the consequences. > > > > > > > > What does "manually go through those mailboxes and convert/delete/add the > > ; the message files so that they appear to have been stored using > > ; your new format list" exactly imply here ? > > It sounds like you have to shutdown Asterisk, find the old voicemail > messages stored on the server in their respective directories, and > convert all of them to the new file/audio format, then modify > voicemail.conf and then re-start Asterisk. At least that's what it > sounds like to me. In other words, it seems that Asterisk will get > confused if it finds old voicemail messages in the storage with a format > different from what voicemail.conf tells it to expect.
You must also remove all message content with a format that is NOT in your new list (i.e. if you remove a format from the list, you must also remove all recordings in the voicemail hierarchy with that format). And really, this is the critical step for most messages. I believe only forwarding messages with prepend would be a problem for not creating new files. > I suppose if you go and delete all the old voicemail messages before > changing the format in voicemail.conf, it will work equally well. > However, your users might not be very pleased :-) That will also work. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- 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
