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<p class=MsoNormal><font size=2 face=Arial><span lang=PT style='font-size:10.0pt;
font-family:Arial'>Hi,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=PT style='font-size:10.0pt;
font-family:Arial'> </span></font><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>I’m
new to Asterisk development, I’m in my first steps...<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> I’m
having problems to find detailed information of some of the internal mechanism
of this great software, I can see that are some effort in place but not yet a
full documentation (which is normal, I would say).<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>What I would like to know is the right way to pass some data
I have on the zap channel side (chan_zap.c) to the SIP channel when the Sip channel
is dialed from a Zap channel. What is the correct mechanism to use? Is it
possible using pbx_builtin_setvar_helper() and pbx_builtin_getvar_helper()
functions?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Thanks for any tip that can push me in the right direction.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>BTW, what I’m trying to do is to accomplish the
following (as I stated on a message on the asterisk.org forum but with no help
so far):</span></font><br>
<br>
<span class=postbody>I'm trying to pass user-user information from a PRI call
setup to a SIP INVITE, and vice versa (Probably I will add the info to the
SDP). </span><br>
<br>
<span class=postbody>I confirmed that the information is available on the
libpri, and I believe it is also available on the chan_zap but have not yet
confirmed this last point at runtime. </span><br>
<br>
<span class=postbody>My question is how I can pass this information on a
bridged call from the Zap channel to the SIP channel. What mechanisms are
available on Asterisk to do this kind of things? </span><br>
<br>
<span class=postbody>Thanks in advance for any help,<o:p></o:p></span></p>
<p class=MsoNormal><span class=postbody><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'> Ricardo</span></font></span><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></p>
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