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<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Here is a question that needs a few
opinions...</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Recently we installed a couple of FXS gateways into
a site with a SIP Proxy/Softswitch other than Asterisk. One of
the things that the users on this site need to do is receive calls on single
line phones on the FXS gateways and then hookflash and transfer them to other
SIP users.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>We found that the FXS units, true to their nature
as VoIP gateways, saw the hookflash and passed a SIP INFO (event hookflash)
back to the Proxy. The Proxy sent this message on to the calling SIP phone
which replied that this "feature is not implemented." </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>The gateway manufacturer says that the Proxy
should process the INFO packet, place the calling endpoint on hold (as a PBX
would), stream dialtone to the gateway prompting the user to dial the digits
indicating the destination to whom the calling party should be transferred, and
then do a transfer.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>The Proxy manufacturer says that the gateway should
see the hookflash, Hold the caller locally (as a SIP phone would), and give
new dialtone to the single line phone prompting the user to dial the digits
digits indicating the destination to whom the calling party should be
transferred, and then send a complete transfer sequence to the
Proxy.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>My question is, how would Asterisk handle a
situation like this? Are there any opinions as to how this scenario should
be handled?</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Sean</FONT> </DIV></BODY></HTML>