[asterisk-bugs] [Asterisk 0018898]: Large number of active sip dialogs PUBLISH in the output "sip show channels".
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Mar 1 13:41:24 CST 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:
======================================================================
Project: Asterisk
Issue ID: 18898
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.8.2.4
JIRA: SWP-3178
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: 2011-03-01 13:41 CST
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Summary: 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.
======================================================================
----------------------------------------------------------------------
(0132498) amilcar (reporter) - 2011-03-01 13:41
https://issues.asterisk.org/view.php?id=18898#c132498
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Folow is a dialog that stick around in the system forever. Asterisk sends
489 "Bad Event", but the sip dialog remains in the system:
(the client, as you can see is X-Lite release 1100l stamp 47546 in this
case, and i've changed the domain name for privacy)
<--- SIP read from UDP:189.39.17.2:33313 --->
PUBLISH sip:3004 at pbx.domain.com SIP/2.0
Via: SIP/2.0/UDP
189.39.17.2:33870;branch=z9hG4bK-d8754z-bd588b0167259706-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:3004 at 189.39.17.2:33313>
To: "Position
https://issues.asterisk.org/view.php?id=4"<sip:3004@pbx.domain.com>
From: "Position
https://issues.asterisk.org/view.php?id=4"<sip:3004@pbx.domain.com>;tag=4347bb3e
Call-ID: Zjg3YzU4ZTM5MWM4OGQ1OTgzNTZhMzk3OWI3ZGY5MjM.
CSeq: 1 PUBLISH
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/pidf+xml
User-Agent: X-Lite release 1100l stamp 47546
Event: presence
Content-Length: 425
<?xml version='1.0' encoding='UTF-8'?><presence
xmlns='urn:ietf:params:xml:ns:pidf'
xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model'
xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid'
xmlns:c='urn:ietf:params:xml:ns:pidf:cipid'
entity='sip:3004 at pabx.vonix.com.br'><tuple
id='t290fce49'><status><basic>open</basic></status></tuple><dm:person
id='t290fce49'><rpid:activities><rpid:unknown/></rpid:activities></dm:person></presence>
<------------->
--- (14 headers 1 lines) ---
<--- Transmitting (no NAT) to 189.39.17.2:33313 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP
189.39.17.2:33870;branch=z9hG4bK-d8754z-bd588b0167259706-1---d8754z-;rport;received=189.39.17.2
From: "Position
https://issues.asterisk.org/view.php?id=4"<sip:3004@pbx.domain.com>;tag=4347bb3e
To: "Position
https://issues.asterisk.org/view.php?id=4"<sip:3004@abx.domain.com>;tag=as2c86fa71
Call-ID: Zjg3YzU4ZTM5MWM4OGQ1OTgzNTZhMzk3OWI3ZGY5MjM.
CSeq: 1 PUBLISH
Server: Vonix PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
Issue History
Date Modified Username Field Change
======================================================================
2011-03-01 13:41 amilcar Note Added: 0132498
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