[asterisk-bugs] [Asterisk 0012994]: Spamming CLI / logs with 'Remote host can't match request BYE to call...'

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12994 
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Reported By:                pabelanger
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12994
Category:                   Channels/chan_sip/General
Reproducibility:            random
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0-beta9 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             07-04-2008 11:33 CDT
Last Modified:              07-18-2008 03:37 CDT
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Summary:                    Spamming CLI / logs with 'Remote host can't match
request BYE to call...'
Description: 
Every 6/7 seconds, our cli is getting spammed (see below).

sip show channel 05b6e9a07669fa0c0a4d88a663b4a2bd at 192.168.20.2
---
  * SIP Call
  Curr. trans. direction:  Outgoing
  Call-ID:                05b6e9a07669fa0c0a4d88a663b4a2bd at 192.168.20.2
  Owner channel ID:       <none>
  Our Codec Capability:   6
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   4
  Joint Codec Capability:   4
  Format:                 0x0 (nothing)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    192.168.20.3:5070
  Received Address:       192.168.20.4:5070
  SIP Transfer mode:      open
  NAT Support:            RFC3581
  Audio IP:               192.168.20.2 (local)
  Our Tag:                as643fb95e
  Their Tag:              as643fb95e
  SIP User agent:         RTCC/3.0.0.0
  Peername:               sv0071iv
  Original uri:           sip:sv0071iv.internal.xxx.on.ca:5070
  Need Destroy:           No
  Last Message:           Tx: BYE
  Promiscuous Redir:      No
  Route:                 
sip:sv0071iv.internal.xxx.on.ca:5070;transport=Tcp;maddr=192.168.20.3
  DTMF Mode:              rfc2833
  SIP Options:            (none)
  Session-Timer:          Inactive

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---------------------------------------------------------------------- 
 vinsik - 07-18-08 03:37  
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I have the same problem on 1.4.20.1.
After a call transfer asterisk seems to think that the call is still
active.

[Jul 18 11:43:26] DEBUG[6568] chan_sip.c: Auto destroying SIP dialog
'264f8c8f4e671fbc2c23c01406898da7 at rt.cuuma.fi'
[Jul 18 11:43:26] DEBUG[6568] chan_sip.c: Finally hanging up channel after
transfer: 264f8c8f4e671fbc2c23c01406898da7 at rt.cuuma.fi
[Jul 18 11:43:26] DEBUG[6568] chan_sip.c: ** SIP timers: Rescheduling
retransmission 2 to 200 ms (t1 100 ms (Retrans id
http://bugs.digium.com/view.php?id=385675))
[Jul 18 11:43:26] DEBUG[6568] chan_sip.c: = No match Their Call ID:
29A399523AED420EBDAEF4A5164D4CE40x5912ee9d Their Tag 1f971383e806 Our tag:
as4d68b2fa
[Jul 18 11:43:26] DEBUG[6568] chan_sip.c: = No match Their Call ID:
iSWmaDkuoIfcLz8rbhvtXONLxP17rM Their Tag 17v9cagtudhc71ej68di Our tag:
as3f8d11e0
[Jul 18 11:43:26] DEBUG[6568] chan_sip.c: = No match Their Call ID:
3c26700fd6d8-di9tlee4j1qc at snom320-000413245816 Their Tag 77lkqcyrot Our
tag: as01e608eb
[Jul 18 11:43:26] DEBUG[6568] chan_sip.c: = Found Their Call ID:
264f8c8f4e671fbc2c23c01406898da7 at rt.cuuma.fi Their Tag as0196568f Our tag:
as0196568f
[Jul 18 11:43:26] DEBUG[6568] chan_sip.c: Stopping retransmission on
'264f8c8f4e671fbc2c23c01406898da7 at rt.cuuma.fi' of Request 221: Match Found
[Jul 18 11:43:26] DEBUG[6568] chan_sip.c: Allocating new SIP dialog for
(No Call-ID) - OPTIONS (No RTP)
[Jul 18 11:43:26] DEBUG[6568] chan_sip.c: = Found Their Call ID:
1712eb8e11ff032a10a0ba6c731e5d99 at rt.cuuma.fi Their Tag  Our tag:
as42cc043d
[Jul 18 11:43:26] DEBUG[6568] chan_sip.c: Stopping retransmission on
'1712eb8e11ff032a10a0ba6c731e5d99 at rt.cuuma.fi' of Request 102: Match Found

This is what debug 9 is giving. 

Reload of module chan_sip.so does not help. 
Reload of asterisk does not help.
Restart of asterisk helps. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-18-08 03:37  vinsik         Note Added: 0090433                          
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