[asterisk-bugs] [Asterisk 0013865]: SIP/TLS enabled - just one call possible - 481 Call/Transaction Does Not Exist
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Apr 28 13:02:43 CDT 2009
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=13865
======================================================================
Reported By: st
Assigned To: mmichelson
======================================================================
Project: Asterisk
Issue ID: 13865
Category: Channels/chan_sip/TCP-TLS
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.6.1-beta1
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2008-11-09 10:03 CST
Last Modified: 2009-04-28 13:02 CDT
======================================================================
Summary: SIP/TLS enabled - just one call possible - 481
Call/Transaction Does Not Exist
Description:
Settup consits of Asterisk 1.6.1-beta1 and Snom320 firmware 7.3.7
TLS is enabled using standard port 5061 for the first example and port
5062 for the second example.
Dialplan is very simple: Answer; Wait(1); Playback(...); Hangup;
The first call of the first example does work but it is not possible to
call the same number again: 481 Call/Transaction Does Not Exist
The first call of the second example has no "BYE" and has to be cancelled
at the phone.
(IMHO a new category chan_sip/TLS should be created)
======================================================================
----------------------------------------------------------------------
(0103885) vrban (reporter) - 2009-04-28 13:02
http://bugs.digium.com/view.php?id=13865#c103885
----------------------------------------------------------------------
tls_port_v5.patch:
added two breaks; into ast_sip_ouraddrfor. otherwise for TCP/TLS we fall
through and get the udpbindaddr if udpbindaddr != 0.0.0.0 in sip.conf
the ast_debug i added in ast_sip_ouraddrfor i offset to right spot.
Issue History
Date Modified Username Field Change
======================================================================
2009-04-28 13:02 vrban Note Added: 0103885
======================================================================
More information about the asterisk-bugs
mailing list