<DIV>Some providers like Iconnect do not accept 2nd INVITES.. so ASTERISK </DIV>
<DIV>must bridge the audio <IMG src="http://us.i1.yimg.com/us.yimg.com/i/mesg/tsmileys2/02.gif"></DIV>
<DIV> </DIV>
<DIV>Perhaps a way for ASTERISK to force the needed rtp/port/ip can be coded.</DIV>
<DIV> </DIV>
<DIV>1. ASTERISK invites a nat'd phone to acquire the correct audio</DIV>
<DIV> </DIV>
<DIV>2. ASTERISK invites Iconnect with the FIXED port values</DIV>
<DIV> </DIV>
<DIV>thus allowing direct RTP from Iconnect to the nat'd sip phone.</DIV>
<DIV> </DIV>
<DIV>It is tempting to hack chan_sip for this feature, as asterisk makes a good </DIV>
<DIV>effort to handle nat'd phones.</DIV>
<DIV> </DIV>
<DIV>or ? </DIV>
<DIV> </DIV>
<DIV>thx</DIV>
<DIV> </DIV><p>
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