<br><font size=2 face="sans-serif">I am creating a new kind of channel module for asterisk. When receiving an incoming call, my "new channel" needs to have access to calling channel.</font>
<br><font size=2 face="sans-serif">More specifically, i need to know if the calling channel type is <b>"SIP"</b> and in that case, the udp port used for rtp packets (local and remote).</font>
<br><font size=2 face="sans-serif">I have looked all over the <b>"ast_channel struct"</b> and the <b>dialed</b> and <b>dialing</b> fields are allways null. any information on this ?</font>
<br><font size=2 face="sans-serif">thanks for any information related to this.</font>