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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Since this information is available in
debug, it is obviously there for the taking and redistribution. Someone more
versed than I will have to give you a real answer. The “Clunky/hack”
way to get it would be a “teed” log read via AGI/AMI.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Scott McNab<br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, February 05, 2009
2:06 AM<br>
<b><span style='font-weight:bold'>To:</span></b>
asterisk-users@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> [asterisk-users]
Configure Asterisk to preserve SIP header?</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>Hello.<br>
<br>
Is it possible to configure Asterisk to preserve specific SIP INVITE headers
when setting up a call?<br>
<br>
Specifically, I have a custom SIP client that sends an additional header in the
INVITE request when originating a call. This is to request that the call is
auto-answered by the destination phone. i.e.<br>
<br>
Call-Info: <sip:192.168.100.50>;answer-after=0<br>
<br>
If I use wireshark to sniff the packets, I can see that this header is present
in the original INVITE request from my SIP phone to Asterisk, however, in the
INVITE message that Asterisk sends to the recipient phone, this header has been
stripped.<br>
<br>
Is it possible to configure Asterisk so that it forwards this SIP header
intact? <br>
<br>
I know that it is possible to set up a dialplan to insert this header for
specific extensions, but I really would like to be able to generate this header
using my client!<br>
<br>
Any ideas would be greatly appreciated!<br>
Thanks<br>
Scott<o:p></o:p></span></font></p>
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