[asterisk-dev] SIP TLS handshake needs a timeout

[Russell Bryant]

David Vossel wrote:
> Perhaps I am overlooking some obvious solution here.  Does anyone have any ideas?

I would suggest joining the openssl-users mailing list and asking for
help there.

http://www.openssl.org/support/community.html

-- 
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org