[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 08:33:15 CDT 2011
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=18898
======================================================================
Reported By: Obi Van
Assigned To:
======================================================================
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:
======================================================================
Date Submitted: 2011-02-28 07:55 CST
Last Modified: 2011-05-24 08:33 CDT
======================================================================
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.
======================================================================
----------------------------------------------------------------------
(0135332) Chainsaw (reporter) - 2011-05-24 08:33
https://issues.asterisk.org/view.php?id=18898#c135332
----------------------------------------------------------------------
Gareth, I have this patch applied but an RX: Publish session from Ekiga has
been open for 3 hours now. What is the expected kill/expiry time for these
please?
Details follow, IPs obfuscated:
pink*CLI> sip show channel
1cde0af7-5b84-e011-91df-001742895bf3 at paulh-laptop2
* SIP Call
Curr. trans. direction: Incoming
Call-ID:
1cde0af7-5b84-e011-91df-001742895bf3 at paulh-laptop2
Owner channel ID: <none>
Our Codec Capability: 0xc7f000c
(ulaw|alaw|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140)
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: 1.2.3.4:5060
Received Address: 1.2.3.4:5060
SIP Transfer mode: open
Force rport: No
Audio IP: 4.3.2.1 (local)
Our Tag: as5a7efd47
Their Tag: f88d0bf7-5b84-e011-91df-001742895bf3
SIP User agent: Ekiga/3.2.7
Need Destroy: No
Last Message: Rx: PUBLISH
Promiscuous Redir: No
Route: N/A
DTMF Mode: rfc2833
SIP Options: (none)
Session-Timer: Inactive
I was expecting it to be:
Sess-Expires : 1800 secs
Issue History
Date Modified Username Field Change
======================================================================
2011-05-24 08:33 Chainsaw Note Added: 0135332
======================================================================
More information about the asterisk-bugs
mailing list