[Asterisk-Users] tiny audio drops (blips)

[Matt Hess]

We are receiving multiple audio drop outs on calls .. I've done quite a 
bit of troubleshooting and it only involves calls that require the 
Dial(SIP/xxx,,t) for transfers.. as long as the media path goes through 
the server the audio blips happen.. using ulaw codec, btw. I have been 
able to align the blips in audio to a specific point involving 
asterisk.. it seems to happen right at about the time asterisk is 
dealing with another call..
ie:     -- Called xxxx at other_proxy

It's really an aggravating thing.. what I am asking is this.. we use sip 
info for dtmf.. (works great for us).. why must the audio stream be 
running through asterisk if sip info is being used? The # still goes to 
the asterisk server.. what is the harm in setting up a fresh call leg 
and reinviting the media end point (party being transferred) over to the 
new call? It's not like asterisk needs to or even does receive the dtmf 
inband if it's using sip info anyway right?

A few pointers would be appreciated as to smoothing asterisk out some so 
that other calls being setup do not affect current calls.



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