[asterisk-users] RTP audio

[Jerry Geis]

Has there been issues where "once in a while" RTP audio does not work ?

Example: connection to Cisco call manager - works mostly all the time.

once in a great while - person does not hear the "beep" when calling in.
once in a great while - person they hear the beep - but do not hear the
audio public address.

What would I be looking for to track this beast down ?

This is my SIP trunk
[LSVOIP]
type=friend
dtmfmode=rfc2833
secret=password
username=LSVOIP
defaultuser=LSVOIP
disallow=all
allow=ulaw
allow=alaw
context=incoming
host=172.1.1.1
canreinvite=yes
qualify=yes
insecure=invite

Thoughts?

Jerry
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