Asterisk can convert from wav but it still needs to be in the correct format. See http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk
Jonas Kellens wrote: > Hello list, > > when putting the class 'default' in comment, then this happens : > > [Aug 13 12:36:34] WARNING[21172]: res_musiconhold.c:666 get_mohbyname: > Music on Hold class 'default' not found > [Aug 13 12:36:34] -- Started music on hold, class '106002', on > SIP/test2-00000001 > [Aug 13 12:36:34] WARNING[21172]: format_wav.c:124 check_header: Does > not say fmt > [Aug 13 12:36:34] WARNING[21172]: file.c:385 fn_wrapper: Unable to open > format wav > [Aug 13 12:36:34] WARNING[21172]: res_musiconhold.c:251 > ast_moh_files_next: Unable to open file > '/var/lib/asterisk/moh/106002/01Long': No such file or directory > > Questions : > > 1. how can I use AND class "default" AND class "106002" ?! > 2. is it normal that Asterisk can not convert from wav to alaw/gsm ?! > > > Jonas. > > > On 08/13/2010 09:57 AM, Jonas Kellens wrote: >> Hello list, >> >> >> I'm using asterisk 1.4.30 and realtime sip. >> >> >> I notice that the field "musiconhold" is not working as when putting >> someone on hold, the default musiconhold class is always used. >> >> >> musiconhold.conf : >> >> [default] >> mode=files >> directory=/var/lib/asterisk/moh >> random=yes >> ; >> [106002] >> mode=files >> directory=/var/lib/asterisk/moh/106002 >> random=yes >> >> >> my realtime sip peers have the following in the column '*musiconhold*' >> : *106002* >> >> >> asterisk*CLI> moh show classes >> Class: default >> Mode: files >> Directory: /var/lib/asterisk/moh >> Class: 106002 >> Mode: files >> Directory: /var/lib/asterisk/moh/106002 >> >> >> But always : >> >> [Aug 13 09:47:57] -- Started music on hold, class 'default', on >> SIP/test2-00000014 >> [Aug 13 09:48:05] -- Stopped music on hold on SIP/test2-00000014 >> >> >> >> Can anyone help ?! >> >> >> Kind regards, >> >> Jonas. >> -- _____________________________________________________________________ -- 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
