[asterisk-users] sometimes calls drop during attended transfer

[gincantalupo]

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/12 at inbound_sip-f8de,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