[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
Thu Apr 23 06:42:06 CDT 2009
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=13865
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Reported By: st
Assigned To: mmichelson
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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:
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Date Submitted: 2008-11-09 10:03 CST
Last Modified: 2009-04-23 06:42 CDT
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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)
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(0103676) vrban (reporter) - 2009-04-23 06:42
http://bugs.digium.com/view.php?id=13865#c103676
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>st_sip_ouraddrfor, you sometimes reference p->ourip. Instead, you should
be using the "us" variable. This way, things are more consistent.
done.
>I am having difficulty finding where req->socket.type is set
req.socket.type is set in: sip_tcp_helper_thread/sipsock_read,
then going to:
handle_request_do ->
find_call ->
sip_alloc ->
ast_sip_ouraddrfor, and here we are.
Issue History
Date Modified Username Field Change
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2009-04-23 06:42 vrban Note Added: 0103676
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