Hi Michael, Can you show us the output from: "moh show classes" and "moh show files" Command
Or try it to set a new exten after setting the language with: exten => 12345,n,Set(CHANNEL(musicclass)=personalised) Daniel Am 13.06.2010 um 12:35 schrieb Mickael Monsieur: > Hello, > The MeetMe application refuses MusicOnHold personalized and skip always in > the default! > Have you any idea how to fix this? > > -- Executing [028883...@default:1] Set("SIP/109.10.214.1-00000002", > "CHANNEL(language)=fr") in new stack > -- Executing [028883...@default:2] Answer("SIP/109.10.214.1-00000002", > "") in new stack > -- Executing [028883...@default:3] Playback("SIP/109.10.214.1-00000002", > "welcome") in new stack > -- <SIP/109.10.214.1-00000002> Playing 'welcome.alaw' (language 'fr') > [Jun 13 12:30:00] NOTICE[13437]: channel.c:3012 __ast_read: Dropping > incompatible voice frame on SIP/109.10.214.1-00000002 of format ulaw since > our native format has changed to 0x8 (alaw) > -- Executing [028883...@default:4] > MeetMeCount("SIP/109.10.214.1-00000002", "100,COUNT") in new stack > == Parsing '/etc/asterisk/meetme.conf': == Found > -- Executing [028883...@default:5] GotoIf("SIP/109.10.214.1-00000002", > "0?100") in new stack > -- Executing [028883...@default:6] MeetMe("SIP/109.10.214.1-00000002", > "100,1pdM(personnalised)") in new stack > -- Created MeetMe conference 1023 for conference '100' > -- Started music on hold, class 'personnalised', on > SIP/109.10.214.1-00000002 > -- Stopped music on hold on SIP/109.10.214.1-00000002 > -- Started music on hold, class 'default', on SIP/109.10.214.1-00000002 > > Thank you, > Mickael. > -- > _____________________________________________________________________ > -- 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 Daniel Knoll Liberdastr. 9 12047 Berlin fon +49 (0)179 20 16 50 8 mail dan...@danielknoll.de web www.danielknoll.de
-- _____________________________________________________________________ -- 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