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<p class=MsoNormal>I have an Asterisk 1.6.2 server on a public IP, Cisco 7940
on the localnet, and a trunk to Sipphone/Gizmo/Google Voice. The
externhost and localnet parameters are all set correctly in sip.conf. An
inbound call from Sipphone works great until the local channel places the call
on hold. During hold, the Sipphone user cannot hear music, only
silence. The silence continues after the hold, though the local phone can
hear the Sipphone user.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Every possible combination of nat=yes, no, maybe, possibly
or never gives the same result. Further, canreinvite=yes/no/nonat has no
result. I suspect a possible reinvite issue with Asterisk being out of
the RTP stream, so I have tried all the usual variables in the DialI() command
as well to no avail.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Any thoughts on how to fix one-way-audio after a hold?<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>--Brent<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
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