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Hi All I have a problem with IVR scripts which consist mainly of Playback of audio files, driven from an AGI application. There are clicks every few seconds or more frequently that is audible on the remote end (PSTN), but not on the Asterisk recording of the call. If I record the remote end and compare it to the local recording, it appears to be about 5%-7% too fast - i.e. if I synchronise the

...r Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. DeadAGI variables confusion (Dave Brooks) 2. SV: [Asterisk-Users] Outbound Caller ID number on E1 (jan.sarin@securia.se) 3. Outbound Call & SIP Results (David Brazier) 4. Re: Outbound Caller ID number on E1 (Steve Underwood) 5. Delaying media stream by short period after 183 is sent (Mark van Kerkwyk) 6. RE: RE: [Asterisk-Users] Blocked Callerid (Alexander Lopez) 7. Anyone know a good ITSP in Canada that supports *? (hugolivude) 8. [Fwd: Re:...

Just checked my figures, and I mean 0.5%-0.7%. Anyway, it is the resulting clicks that are the problem. Any help still appreciated. David -----Original Message----- From: David Brazier Sent: 13 March 2007 00:33 To: asterisk-users@lists.digium.com Subject: Playback 5% Too Fast? Hi All I have a problem with IVR scripts which consist mainly of Playback of audio files, driven from an AGI application. There are clicks every few seconds or more frequently that is audible on the rem...

David Brazier wrote: > Hi > > The latest SVN trunk for speex has changed the SpeexPreprocessState to > an opaque structure, for jolly good software engineering reasons. > However, the Analogue AGC (AAGC) feature of iaxclient (in audio_enode.c) > relies on some members of this. It uses speech_p...

Hi The latest SVN trunk for speex has changed the SpeexPreprocessState to an opaque structure, for jolly good software engineering reasons. However, the Analogue AGC (AAGC) feature of iaxclient (in audio_enode.c) relies on some members of this. It uses speech_prob to detect when there is enough speech to consider AAGC and then loudness2 to decide how to adjust the input mixer. We want to use

Hi My VoIP provider demands that the address on the INVITE line includes the port number, so: INVITE sip:12345@1.2.3.4:5060 SIP/2.0 Not: INVITE sip:12345@1.2.3.4 SIP/2.0 I can see example logs from other users where this appears to have been done, but I can't find how to do it. Any suggestions? I apologise if the answer is obvious! David