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<DIV><FONT face=Arial size=2>I'm having trouble with both my Polycom IP600 and
IP500 disconnecting calls to the PSTN after about 1 hour. The below log is of a
phone call that lasted 1hr 39mins which is my record so far. I cannot figure out
what is causing the call to terminate and I am hoping somone on this list can
help me. In this example both the phone and the asterisk server have public IP
addresses so NAT shoul not be an issue whatsoever. Any ideas or help would be
greatly appreciated.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>-DevilFish</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Asterisk Version: </FONT><FONT face=Arial
size=2>Asterisk CVS-v1-0-01/13/05</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Call Start:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Jan 15 12:46:59
VERBOSE[4290]: -- Executing SetGroup("SIP/302-928e",
"302") in new stack<BR>Jan 15 12:46:59 VERBOSE[4290]: --
Executing Dial("SIP/302-928e", "<A
href="mailto:SIP/12699264242@sip_proxy-out|30">SIP/12699264242@sip_proxy-out|30</A>")
in new stack<BR>Jan 15 12:46:59 VERBOSE[4290]: --
SIP/sip_proxy-out-f201 is making progress passing it to SIP/302-928e<BR>Jan 15
12:47:08 VERBOSE[4290]: -- SIP/sip_proxy-out-f201
answered SIP/302-928e<BR>Jan 15 12:47:08 VERBOSE[4290]:
-- Attempting native bridge of SIP/302-928e and
SIP/sip_proxy-out-f201<BR></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Call Terminating</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Jan 15 14:25:42 DEBUG[4290]: Stopping
retransmission on <A
href="mailto:'1a59ffd871845b5d27c9b56824a47d5b@voip.acd.net'">'1a59ffd871845b5d27c9b56824a47d5b@voip.acd.net'</A>
of Response 661161239: Found<BR>Jan 15 14:25:57 DEBUG[4290]: Stopping
retransmission on <A
href="mailto:'1a59ffd871845b5d27c9b56824a47d5b@voip.acd.net'">'1a59ffd871845b5d27c9b56824a47d5b@voip.acd.net'</A>
of Response 661161240: Found<BR>Jan 15 14:26:15 DEBUG[4290]: Failed to grab
lock, trying again...<BR>Jan 15 14:26:15 VERBOSE[4290]:
-- Started music on hold, class 'default', on SIP/302-928e<BR>Jan 15 14:26:16
DEBUG[4290]: Bridge stops because we're zombie or need a soft hangup:
c0=SIP/302-928e, c1=SIP/sip_proxy-out-f201, flags: No,No,No,Yes<BR>Jan 15
14:26:16 DEBUG[4290]: Bridge stops bridging channels SIP/302-928e and
SIP/sip_proxy-out-f201<BR>Jan 15 14:26:16 DEBUG[4290]: Ignoring too old packet
packet 661161242 (expecting >= 661161243)<BR>Jan 15 14:26:16 DEBUG[4290]:
update_user_counter(12699264242) - decrement inUse counter<BR>Jan 15 14:26:16
DEBUG[4290]: 12699264242 is not a local user<BR>Jan 15 14:26:16 DEBUG[4290]:
Exiting with DIALSTATUS=ANSWER.<BR></FONT></DIV>
<DIV><FONT face=Arial size=2>Call length 1hr 39mins</DIV></FONT></BODY></HTML>