search for: ast_feature_request_and_dial

Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the "transferer". Is there any possibility to recover the call before the timeout of 15 seconds expires? I mean, I would like

...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, We are trying to use attended transfer with Asterisk 1.2.5, but when we do the transfer and dial the new number, it times out after 3 rings and then the callee is put back to the original agent. Where can I adjust the timeout which applies to the number we are transferring to? I have changed the extension for this number to timeout at 60 seconds, but that seems to make no difference. --