[asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds
Andrew Martin
amartin at xes-inc.com
Fri May 8 17:12:28 CDT 2015
Hello,
I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All
the SIP clients are on a LAN, so no NAT is involved. I have been experiencing
an intermittent problem where a call will be successfully answered, but then
dropped by Asterisk 32 seconds after it is answered (with a "Retransmission
timeout reached on transmission" error). Here is an example of this happening
in the asterisk console:
http://pastebin.com/7LDwHAJe
This problem only happens a fraction of the time, so I have been unable to
enable SIP debugging before it happens to get a capture. However, usually the
caller will just call back immediately and then the call will work without a
problem. It sounds like SIP Timer B is what causes the call to be dropped if an
ACK to the INVITE is not received within 32 seconds. How can I determine if
this is the case and how can I resolve this "Retransmission timeout" problem?
Here is my sip.conf:
general]
directmedia=no
directrtpsetup=no
dtmfmode=rfc2833
context=internal
allowsubscribe=no
qualify=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
localnet=10.10.32.0/255.255.248.0
[123]
secret=111111
host=dynamic
type=friend
Thanks!
Andrew Martin
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