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<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'>Hey everyone,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'>I’ve been poking at the chan_sip.c for a little
while now, in an attempt to implement a first prototype of a SIP nat proxy
hack.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'>Up to date, I’ve got my asterisk to register on
my SIP provider (that was the easy part) through their NAT proxy, since the
registration procedure was self-contained and pretty-much simple.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'>However, I’m attempting to implement the same
hack (using an alternate hostname/port to send the IP packets without altering
SIP messages) for outbound calls, however, I have yet to identify the source in
the code where the SIP message is actually transmitted, or, where the destination
host/peer are defined to then be delivered. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'>For a while, I was lead to the impression that is was
in the create_addr() function, however, that function only seems assembles the
sip message, while copying data from the possibly found peer to the pvt
structure.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'>Anyone have some pointers, or a description of the
process when a call is initiated from asterisk towards an outbound SIP peer
defined in the sip.conf ? All I want is to send the SIP messages to an
alternate host/port rather than the ones defined in the host= and port=
parameters of the sip.conf<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'>Thanks<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'>Kris<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'>Ps: My C is functional, but not great; I’ve succumbed
too much to framework-based languages.<o:p></o:p></span></font></p>
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