[asterisk-users] sometimes calls drop during attended transfer

[Maxi]

2007/7/5, gincantalupo <gincantalupo at fgasoftware.com>:
> 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
>

Hi Giorgio,

I'm trying to resolv this problem. I have the same situations.

Can you resolve this?

Max

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