[asterisk-users] MixMonitor inserting extra 20ms packets of silence (1.4.43)
I'm trying to diagnose potential causes of an issue with MixMonitor in 1.4.43 (I doubt it's very version-specific). I won't have hands on the kit until the end of the week, but I have listened to some recordings. It doesn't happen on every call - only sometimes. Basically, it is a call from one SIP phone extension to another, and the dialplan sets up MixMonitor on the calling channel before doing the Dial. The call sounds fine to both parties in the call, but when listening back to the recording, the speech is slowed down and broken - a bit like a robot or dalek voice. Examining the recorded audio in Goldwave I can see that every 20ms (the size of a SIP packet), an extra 20ms of silence is being inserted in the recording file. This happens most of the time, punctuated by occasional bursts of clear audio before it starts happening again. Has anyone seen this kind of thing before? Better still, seen it and solved it? The SIP phones are actually Soundwin ATAs. There is no zaptel or dahdi timing source in the system. Are there any known issues with MixMonitor that could cause this behaviour? Any pointers would be appreciated - thanks! Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor inserting extra 20ms packets of silence (1.4.43)
2012-09-10 18:13, Tony Mountifield skrev: I'm trying to diagnose potential causes of an issue with MixMonitor in 1.4.43 (I doubt it's very version-specific). I won't have hands on the kit until the end of the week, but I have listened to some recordings. It doesn't happen on every call - only sometimes. Basically, it is a call from one SIP phone extension to another, and the dialplan sets up MixMonitor on the calling channel before doing the Dial. The call sounds fine to both parties in the call, but when listening back to the recording, the speech is slowed down and broken - a bit like a robot or dalek voice. Examining the recorded audio in Goldwave I can see that every 20ms (the size of a SIP packet), an extra 20ms of silence is being inserted in the recording file. This happens most of the time, punctuated by occasional bursts of clear audio before it starts happening again. Has anyone seen this kind of thing before? Better still, seen it and solved it? The SIP phones are actually Soundwin ATAs. There is no zaptel or dahdi timing source in the system. Are there any known issues with MixMonitor that could cause this behaviour? Any pointers would be appreciated - thanks! Tony Maybe a I'm reaching here but.. I had some very strange issues with broken quality with Monitor and a NFS mount. This was 1.4, but several years ago. I ended up not using NFS in the end. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
