Summary:ASTERISK-00673: [workaround] first ringback when forwarded to a SIP extension is jerky
Reporter:glepine (glepine)Labels:
Date Opened:2003-12-17 10:09:28.000-0600Date Closed:2011-06-07 14:10:40
Versions:Frequency of
Description:A Mediatrix 1102 SIP endpoint calls Asterisk and enters the IVR.  It then enters the digits the call an extension, being the other port of the Mediatrix 1102.

The first ringback heard is very jerky and almost not audible.  Subsequent ringbacks are loud and clear.


The call is made in G.711 u-law.
Comments:By: John Todd (jtodd) 2003-12-21 09:33:28.000-0600

Where is the ringback tone being generated?  By the mediatrix, or by Asterisk's "r" function in Dial?  Do you know how to determine that?  (i.e.: tethereal)  If the mediatrix is doing that ring generation there's nothing * can do about it.  If Asterisk is making the tones, then it's still the mediatrix's problem, since it seems that media setup isn't fast enough on the mediatrix to catch and translate the first few milliseconds of the audio stream.  7960's work fine for the first ring, even if * is producing the audio.

By: glepine (glepine) 2003-12-22 10:13:53.000-0600

I am pretty sure the tone is generated by Asterisk and sent through RTP to the Mediatrix box.  I worked around this problem by disabling the comfort noise (RTP payload type 13) on the caller side (the Mediatrix box).

I am not sure what it corrects, though.

By: jrollyson (jrollyson) 2004-01-12 00:45:03.000-0600

Anyone able to confirm this?

By: jrollyson (jrollyson) 2004-01-14 03:52:59.000-0600

Doh, comfort noise isn't properly supported by Asterisk.

By: jrollyson (jrollyson) 2004-01-14 03:54:14.000-0600

This isn't really a bug, since asterisk doesn't claim to support comfort noise.