search for: __ast_smoother_feed

Displaying 5 results from an estimated 5 matches for "__ast_smoother_feed".

2005 Jun 10
0
Dropping Frame of G729
Here is the setup: Phone -SIP G729-> AsteriskA -IAX G729-> AsteriskB -SIP G729-> Carrier The call completes but AsteriskA prints on the screen a ton of those "Dropping Frame of G729" messages starting about 5 seconds into the call: Jun 10 11:17:14 NOTICE[14277]: frame.c:135 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Jun 10 11:17:14 NOTICE[14277]: frame.c:135 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Jun 10 11:17:14 NOTICE[14277]: frame.c:135 __ast_smoother_feed: Dropping extra fr...
2009 Jan 05
0
G729 VAD issue
Hi, My setup is SIP Call-->Asterisk-->VSP1 or VSP2 or VSP3 I'm experiencing an interconnect issue with one of the VSP's that seems to have to do with Asterisk not having any VAD control. The error is: NOTICE[10989]: frame.c:203 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end The VSP has switched off silence suppression on their Quitnum device. Any ideas are most welcome. Thanks Shaun
2009 Jan 06
0
G.729 VAD issue
Hi, My setup is SIP Call-->Asterisk-->VSP1 or VSP2 or VSP3 I'm experiencing an interconnect issue with one of the VSP's that seems to have to do with Asterisk not having any VAD control. The error is: NOTICE[10989]: frame.c:203 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end The VSP has switched off silence suppression on their Quitnum device. Any ideas are most welcome. Thanks Shaun
2009 Apr 26
3
Digium fax force T38?
Is it possible to force T38 for all invocations ReceiveFAX() ? Receiving fax always worked OK on Callweaver though I could put SipT38Switchover() into the dial plan. I can't with Digium fax, and it always fails at the point it decides to switch to T38.
2003 Jul 11
8
G729 codec problems
Hi I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator