|Summary:||ASTERISK-00673: [workaround] first ringback when forwarded to a SIP extension is jerky|
|Date Opened:||2003-12-17 10:09:28.000-0600||Date Closed:||2011-06-07 14:10:40|
|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.
****** ADDITIONAL INFORMATION ******
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.