gincantalupo
2007-Jul-05 11:08 UTC
[asterisk-users] sometimes calls drop during attended transfer
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
Noah Miller
2007-Jul-05 13:28 UTC
[asterisk-users] sometimes calls drop during attended transfer
Hi Giorgio -> 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:Some questions: 1. What asterisk version are you using? 2. What are your SIP devices? 3. Who is your SIP provider? (Judging by your CLI output, I'm guessing you're using one.) - Noah
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> _______________________________________________ > --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 >