[Asterisk-Dev] sip call setup

[mark spowage]

Some providers like Iconnect do not accept 2nd INVITES.. so ASTERISK 
must bridge the audio  
 
Perhaps a way for ASTERISK to force the needed rtp/port/ip can be coded.
 
1. ASTERISK invites a nat'd phone to acquire the correct audio
 
2. ASTERISK invites Iconnect with the FIXED port values
 
thus allowing direct RTP from Iconnect to the nat'd sip phone.
 
It is tempting to hack chan_sip for this feature, as asterisk makes a good 
effort to handle nat'd phones.
 
or ? 
 
thx
 

		
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