[asterisk-bugs] [Asterisk 0018898]: [patch] Large number of active sip dialogs PUBLISH in the output "sip show channels".
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue May 24 17:39:39 CDT 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=18898
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Reported By: Obi Van
Assigned To:
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Project: Asterisk
Issue ID: 18898
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: acknowledged
Asterisk Version: 1.8.2.4
JIRA: SWP-3194
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2011-02-28 07:55 CST
Last Modified: 2011-05-24 17:39 CDT
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Summary: [patch] Large number of active sip dialogs PUBLISH
in the output "sip show channels".
Description:
On Debian 5.0 and Asterisk 1.8.2.4 (also 1.8.2.3) in the output "sip show
channels" I see the following (IP addresses is fake):
123.45.678.910 (None) OTMxYmI3YWRjOGN 0x0 (nothing) No
Rx: PUBLISH <guest>
123.45.678.910 (None) ZmNmNmRhMTUyNDQ 0x0 (nothing) No
Rx: PUBLISH <guest>
123.45.678.910 (None) NTZiYjhlNGRlNzY 0x0 (nothing) No
Rx: PUBLISH <guest>
To these addresses are registered softphones clients. Execute a command
"sip show channel" on any of the PUBLISH dialogues gives the following
results (123.45.678.900 - is address Asterisk):
*CLI>sip show channel NTZiYjhlNGRlNzY
* SIP Call
Curr. trans. direction: Incoming
Call-ID: NTZiYjhlNGRlNzY1NGIxMTBhMzFiMTgxNTlkNGNjNmU.
Owner channel ID: <none>
Our Codec Capability: 0x10d (g723|ulaw|alaw|g729)
Non-Codec Capability (DTMF): 1
Their Codec Capability: 0x0 (nothing)
Joint Codec Capability: 0x0 (nothing)
Format: 0x0 (nothing)
T.38 support No
Video support No
MaxCallBR: 0 kbps
Theoretical Address: 123.45.678.910:5060
Received Address: 123.45.678.910:5060
SIP Transfer mode: open
Force rport: Yes
Audio IP: 123.45.678.900 (local)
Our Tag: as2b4a5359
Their Tag: 27384736
SIP User agent: Zoiper rev.6313
Need Destroy: No
Last Message: Rx: PUBLISH
Promiscuous Redir: No
Route: N/A
DTMF Mode: rfc2833
SIP Options: (none)
Session-Timer: Inactive
Number of such dialogues can reach up to 100 or more! CLI command "sip
reload" does not help. Only helps "core stop now". I noticed that
Session-Timer: Inactive
With the execution of commands for any active dialogue, for example ACK, I
get the following:
Session-Timer: Active
S-Timer Interval: 600
S-Timer Refresher: uas
S-Timer Expirys: 0
S-Timer Sched Id: 162202
S-Timer Peer Sts: Inactive
S-Timer Cached Min-SE: 0
S-Timer Cached SE: 600
S-Timer Cached Ref: auto
S-Timer Cached Mode: Originate
While the output is consistent with the settings in the file sip.conf. It
is seen that:
Session-Timer: Active
It seems to me that the dialogue PIBLISH does not work Session-Timer.
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(0135356) gareth (reporter) - 2011-05-24 17:39
https://issues.asterisk.org/view.php?id=18898#c135356
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Chainsaw: maximum expiry time is whatever maxexpiry is set to but could be
less depending on what the client sends in the Expires header.
The session timer settings do not have any effect on publish requests.
There was a bug in the patch, it didn't handle invalid/unknown publish
types, new patch (svn-320715-bad-event.diff) should fix that.
Issue History
Date Modified Username Field Change
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2011-05-24 17:39 gareth Note Added: 0135356
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