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<DIV><FONT face=Arial size=2>Hello All!</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I am in the process of assembling an asterisk-based
phone system for my office - using SPA-3000s to connect the network
to the PSTN. I am wondering if anybody else can get (or has
already seen) the same behaviour out of their 3000.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>The short version: Send multiple Calls
to the SPA's FXO port at the same time it is re-registering with
Asterisk.</FONT></DIV>
<DIV><FONT face=Arial size=2> SPA HTTP
Configuration: PSTN Line -> Register
Expires: 5 (to
ensure it is registering all the time)</FONT></DIV>
<DIV><FONT face=Arial size=2> Dial one number through the
SPA's FXO port - establish a conversation</FONT></DIV>
<DIV><FONT face=Arial size=2> Dial another number through the
same FXO port (SPA3000/NXXXXXY).</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>What SHOULD happen is the second caller receives a
'504 - Service Unavailable' error while the first caller happily continues the
established conversation. What happens here: the
already established call gets dropped, AND the second caller gets a 504
error.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I did send a note to Linksys - and will see what
kind of reponse they have.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>With longer "Register Expires:" times (10, 30, 60
seconds) it took more attempts to make the call
drop. </FONT></DIV>
<DIV><FONT face=Arial size=2>I have my Register Expires time cranked up to 86400
(1 day) now - and am hoping I don't see another repeat. </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>-------------------</FONT></DIV>
<DIV><FONT face=Arial size=2>There are three SPA-3000s in the
system. I looked at some more complicated
dialplan layouts, and decided to keep it simple:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>exten => s,1,Dial(${PSTN2}/${ARG1},,n)<BR>exten
=> s,2,Dial(${PSTN3}/${ARG1},,n)<BR>exten =>
s,3,Dial(${PSTN1}/${ARG1},,n)<BR>exten => s,4,Wait(1)<BR>exten =>
s,5,Playback(all-circuits-busy-now)<BR>exten =>
s,6,Congestion()<BR></DIV></FONT>
<DIV><FONT face=Arial size=2>PSTN1,2,3 are 3 SPA-3000s registered with
Asterisk.</FONT></DIV>
<DIV><FONT face=Arial size=2>This approach relies on the SPA denying a call if
it is already in use.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV>
<DIV><FONT face=Arial size=2>Looking through the logs, the SIP packets
seem to be in order. INVITE, 100-Trying, 504-Service
Unavailable, ACK.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV></DIV>
<DIV><FONT face=Arial size=2>I'm at the end of my technical limit (ever
increasing as I play in the open-source world) - but my best guess
is:</FONT></DIV>
<DIV><FONT face=Arial size=2>During the Register process, something is
temporarily reset (such as a variable indicating that the line is in use)
such that when the second call comes in - it is actually connected to the
existing conversation for a brief period before the SPA realizes that the line
is actually already in use. As part of a cleanup
procedure - a hangup procedure is run: disconnecting the call.
</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>The Equipment my trials were done on:</FONT></DIV>
<DIV><FONT face=Arial size=2>SPA3000 Hardware Version:
2.0.1(7376), Software Version: 3.1.10(GWd), and
also tried Software 3.1.7. </FONT></DIV>
<DIV><FONT face=Arial size=2>Nothing plugged into the FXS
port. </FONT></DIV>
<DIV><FONT face=Arial size=2>Asterisk 1.2.4 running on FreeBSD 5.4
(i386), AMD Athlon 64 3200+, 1GB RAM.</FONT></DIV>
<DIV><FONT face=Arial size=2>SNOM 320. Application-Version: snom320-SIP
5.3.6 Rootfs: snom320 jffs2 v3.36</FONT></DIV>
<DIV><FONT face=Arial size=2>Polycom IP501 <don't have access to the
software/hardware version from where I am right now></FONT></DIV>
<DIV><FONT face=Arial size=2>Cellphone</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>All SIP equipment is running on a dedicated
LAN. Network "splitters" were used to run two parallel LANs through the
existing cabling. (cat5e has 4 twisted pairs, only 2 twisted pairs
are needed for a 100BASET connection) The only computers on the LAN are the
asterisk box, and my workstation (2 NICs each). </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Regards,</FONT></DIV>
<DIV><FONT face=Arial size=2>Dana Harding</FONT></DIV></BODY></HTML>