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@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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