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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I am using SER as a proxy, and using Asterisk as a PBX. A
user calls in to a 1-800 number. They listen to the IVR on one Asterisk PBX,
and then are transferred to the call center at the other Asterisk PBX. Calls
are being brought into the system via SIP. I need to transfer users from one
Asterisk box to the other. Functionally this works fine, practically it doesn’t
as Asterisk forces the RTP stream to go through the first box into the second.
That kills latency and makes the calls unusable. Has anyone else had a similar
problem? I’ve been looking for a while, and am now fairly experienced
with Asterisk. Is there a way I don’t know of to get Asterisk to do the
SIP call transfer? Is there a way I can signal back to the SER proxy not to
hang up the call but to transfer it if I can’t get Asterisk do what I
want without hacking it?<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 perfectly capable of adding this functionality to
Asterisk if necessary, I just don’t want to spend the time if there is
already a way to do this. Maybe I’m doing something stupid and don’t
realize it.<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!<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'>Robert<o:p></o:p></span></font></p>
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