[asterisk-bugs] [Asterisk 0018898]: Large number of active sip dialogs PUBLISH in the output "sip show channels".

[Asterisk Bug Tracker]

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:                   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:              
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Date Submitted:             2011-02-28 07:55 CST
Last Modified:              2011-02-28 18:01 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.

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---------------------------------------------------------------------- 
 (0132461) isrl (reporter) - 2011-02-28 18:01
 https://issues.asterisk.org/view.php?id=18898#c132461 
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i had them on snoms and turned off publish presence which made them go
away
just wondering if that affects native call completion of the snoms 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-28 18:01 isrl           Note Added: 0132461                          
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