[asterisk-users] sip reload and TCP transport.

[Danny Nicholas]

Try changing qualify=yes to qualify=90 on your TCP peers.

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Friday, January 27, 2012 9:45 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] sip reload and TCP transport.

 

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

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