[asterisk-users] sip reload and TCP transport.

[Bryant Zimmerman]

I am running into an interesting issue with asterisk 1.8.x

When I do a sip reload any peer that is using TCP as the transport protocol 
losses it Host entry from the reg list. and it goes to UNKNOW in the Status 
col.  Endpoints using UDP transport do not do this.  This means that for 3 
to 5 min after a sip reload the TCP transport endpoints can't be called. 
Any ideas on what could cause this and how I could fix it. 

Thanks

Bryant
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120127/5c3c533a/attachment.htm>