[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-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:              2011-03-02 08:27 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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---------------------------------------------------------------------- 
 (0132520) amilcar (reporter) - 2011-03-02 08:27
 https://issues.asterisk.org/view.php?id=18898#c132520 
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An example of another endpoint (this time a snom IP phone, useragent
snom190/3.60x):


<--- SIP read from UDP:192.168.0.245:2051 --->
PUBLISH sip:0702 at 192.168.0.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.245:2051;branch=z9hG4bK-g5t37zvtlwta;rport
From: "Lauriane" <sip:0702 at 192.168.0.1>;tag=wslu8n69as
To: "Lauriane" <sip:0702 at 192.168.0.1>
Call-ID: 3c2e08c6a875-104w99db1fdq at snom190
CSeq: 1 PUBLISH
Max-Forwards: 70
Event: number-guessing
Content-Type: application/text
Content-Length: 27

Number: 0715
Max-Hits: 3
<------------->
--- (10 headers 2 lines) ---

<--- Transmitting (no NAT) to 192.168.0.245:2051 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP
192.168.0.245:2051;branch=z9hG4bK-g5t37zvtlwta;rport;received=192.168.0.245
From: "Lauriane" <sip:0702 at 192.168.0.1>;tag=wslu8n69as
To: "Lauriane" <sip:0702 at 192.168.0.1>;tag=as0023ecea
Call-ID: 3c2e08c6a875-104w99db1fdq at snom190
CSeq: 1 PUBLISH
Server: Vonix PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

This sip dialog get stuck too. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-03-02 08:27 amilcar        Note Added: 0132520                          
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