2007/7/5, gincantalupo <[EMAIL PROTECTED]>: > Hi, > I'm testing attended transfer with 3 SIP phones. I noticed about 10% of > my transfers make the call drop and I get this on my log: > Jul 5 10:42:32 WARNING[23960]: file.c:592 ast_readaudio_callback: > Failed to write frame > -- Playing 'beep' (language 'it') > Jul 5 10:42:32 WARNING[23960]: res_features.c:745 builtin_atxfer: > Failed to play transfer sound! > > Moreover, every time I try to transfer from called phone to a third > phone I get this message: > > -- SIP/5-082a9f78 answered Local/[EMAIL PROTECTED],2 > Jul 5 13:02:40 NOTICE[24701]: res_features.c:1171 > ast_feature_request_and_dial: Don't know what to do about control frame: -1 > > > Is there anybody experiencing this problem? Searched on internet without > success. > > TIA > > Giorgio >
Hi Giorgio, I'm trying to resolv this problem. I have the same situations. Can you resolve this? Max > _______________________________________________ > --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 > _______________________________________________ --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
