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<DIV><SPAN class=165240907-23082005><FONT face=Arial size=2>For canreinvite=yes
to work, I think I need to remove the t argument in the Dial(SIP/ext|60|t)
application. Otherwise, Asterisk will allways stay in the middle. I don't want
that, so I removed the 't' argument. That works. Now, when two UA are calling,
Asterisk gets out of the RTP stream. However, when removing the 't' argument,
the Music On Hold doesn't work anymore between these two UA. If I put one UA on
hold, Asterisk states that it is starting Music On Hold, but the holding party
doesn't hear the audio stream.</FONT></SPAN></DIV>
<DIV><SPAN class=165240907-23082005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=165240907-23082005><FONT face=Arial size=2>Is this
resolvable?</FONT></SPAN></DIV>
<DIV><SPAN class=165240907-23082005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=165240907-23082005><FONT face=Arial
size=2>Thanks,</FONT></SPAN></DIV>
<DIV><SPAN class=165240907-23082005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=165240907-23082005><FONT face=Arial size=2>Ronald
Voermans</FONT></SPAN></DIV></BODY></HTML>