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<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2>All,</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial size=2>just started using
SER (great experience so far!). I'm trying to us it for a contact centre
application where I have a specific requirement that calls can only be
terminated in one direction, from a Cisco AS53xx gateway, not a Cisco
phone.</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial size=2>Here's the set-up
and desired result:</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial size=2>User logs into the
contact centre systems which initiates a call on the PSTN side of the AS53xx.
This then creates a VOIP (SIP) call leg from a dial-peer on the gateway, via the
SER proxy.</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial size=2>The agents all have
Cisco 7960 IP phones which register with the SER registrar. The INVITE is
proxied via a simple location lookup in the SER route[0] config file to the
appropriate handset. The agent answers the call and is "logged" in to the
system.</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial size=2>Now's the fun part.
I want to prevent the SIP leg being torn-down by the agent either accidentally
or on purpose hanging up the call from the handset.</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial size=2>Here's the logic
(pseudo code) I have though out: Trap the BYE from the handset using a
statement something like:</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=765425118-23082004>
<FONT face=Arial size=2>if (method=="BYE" and uri=~"sip:[0-9]+@*")
{</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004>
<FONT face=Arial size=2># Agent phone initiated the BYE so do
something</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004>
<FONT face=Arial size=2>route(1);</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004>
<FONT face=Arial size=2>} else {</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2> #do something else - pass on
the message I guess</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2> };</FONT></SPAN></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial size=2>route[1]
{</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004> <FONT face=Arial size=2>#
use exec to call an external function</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004> <FONT face=Arial
size=2>exec_msg ('script.file');</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004> <FONT face=Arial
size=2>}</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial size=2>The external script
needs to OK the BYE message - I assume use the FIFO function to construct the OK
message. </FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial size=2>Now the tricky part:
how do I stop the called phone from hanging up, (I tried a simple 403 Forbidden
response) but that didn't work. So I thought about the REFER method with the URL
of the phone sending the BYE message- can I send a refer to the Gateway (AS5300)
that still thinks the dialogue is open to the phone and "hopefully" get it to
send a new INVITE dialogue to the phone? Or would I have to generate a new
INVITE sequence from the SER proxy exec code? </FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial size=2>If its the latter,
then I need access to the RTP port number on the AS53xx from the original INVITE
message - and this implies access to the SDP payload - which I can't see from
the SER script (unless anyone can tell me how?)</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial size=2>Thanks in
advance!!!</FONT></SPAN></DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=765425118-23082004><FONT face=Arial
size=2>Neill....:o)</FONT></SPAN></DIV>
<P style="MARGIN-TOP: 0px; MARGIN-BOTTOM: 0px" align=left><FONT
face="MS Sans Serif" color=#ff8000>Neill Wilkinson</FONT></P>
<P style="MARGIN-TOP: 0px; MARGIN-BOTTOM: 0px" align=left><FONT
face="MS Sans Serif" color=#ff8000>Senior Consultant</FONT></P>
<P style="MARGIN-TOP: 0px; MARGIN-BOTTOM: 0px" align=left><FONT
face="MS Sans Serif" color=#ff8000>Quortex Consultants Ltd.</FONT></P>
<P style="MARGIN-TOP: 0px; MARGIN-BOTTOM: 0px" align=left><FONT
face="MS Sans Serif" color=#ff8000>e: <A
href="mailto:neill.wilkinson@quortex.com">neill.wilkinson@quortex.com</A></FONT></P>
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