[Asterisk-bugs] [Asterisk 0010107]: A permanent INVITE that gets BUSY a SIP user

noreply at bugs.digium.com noreply at bugs.digium.com
Wed Jul 4 09:49:06 CDT 2007


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=10107 
====================================================================== 
Reported By:                ibc
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   10107
Category:                   Channels/chan_sip/General
Reproducibility:            sometimes
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.4 
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        No 
Request Review:              
====================================================================== 
Date Submitted:             07-03-2007 02:59 CDT
Last Modified:              07-04-2007 09:49 CDT
====================================================================== 
Summary:                    A permanent INVITE that gets BUSY a SIP user
Description: 
I have a queue (with no agents) all of the members using SJphone for
Windows and configured with "call-limit=1".

Sometimes one of them (not always the same user) can't receive calls.

For example, I capture the error now that it's occurring to a SIP username
"lluc-softphone" with extension 209:

-----------------------------------------------------
asterisk*CLI> dial 209
-- Executing [209 at desde-usuarios:1] Dial("OSS/dsp",
"SIP/lluc-softphone|600|wtWT") in new stack
[Jul  3 10:01:22] ERROR[30342]: chan_sip.c:3060 update_call_counter: Call
to peer 'lluc-softphone' rejected due to usage limit of 1
    -- Couldn't call lluc-softphone
  == Everyone is busy/congested at this time (0:0/0/0)
  == Auto fallthrough, channel 'OSS/dsp' status is 'CHANUNAVAIL'
-----------------------------------------------------


But that's not true, in fact:

-----------------------------------------------------
asterisk*CLI> show channels
Channel              Location             State   Application(Data)
0 active channels
0 active call
-----------------------------------------------------


The queue where SJphone users are members is called "interactiva" and the
most strange thing is the following:

-----------------------------------------------------
asterisk*CLI> queue show interactiva
interactiva  has 0 calls (max unlimited) in 'rrmemory' strategy (8s
holdtime), W:0, C:16, A:3, SL:0.0% within 0s
   Members:
      SIP/wadrian-softphone (Not in use) has taken 5 calls (last was 1308
secs ago)
      SIP/lluc-softphone (Busy) has taken no calls yet
      SIP/goni-softphone (Unavailable) has taken no calls yet
      SIP/jose-softphone (Unavailable) has taken no calls yet
      SIP/pep-softphone (Unavailable) has taken 5 calls (last was 331311
secs ago)
      SIP/todiaz-softphone (Not in use) has taken 6 calls (last was 331572
secs ago)
   No Callers
-----------------------------------------------------

As you see SIP/lluc-softphone appears as BUSY.

And more strange:

-----------------------------------------------------
asterisk*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     Last
Message
192.168.1.116    (None)      2084351917@  00101/45278  unkn  No       Rx:
REGISTER
192.168.1.63     (None)      DE39D9DA-0F  00101/00090  unkn  No       Rx:
REGISTER
192.168.1.61     (None)      0D7FEDFE-97  00101/00028  unkn  No       Rx:
REGISTER
192.168.1.65     (None)      C05B1336-1D  00101/00096  unkn  No       Rx:
REGISTER
192.168.1.61     (None)      974C923B-50  00101/00076  unkn  No       Rx:
OPTIONS
192.168.1.63     (None)      4518C18D-AC  00101/00259  unkn  No       Rx:
OPTIONS
192.168.1.64     (None)      9F38E4D6-6A  00101/00276  unkn  No       Rx:
OPTIONS
192.168.1.65     (None)      CC7C13B8-1D  00101/00279  unkn  No       Rx:
OPTIONS
192.168.1.73     (None)      C1E4F0BA-E6  00101/00225  unkn  No       Rx:
OPTIONS
192.168.1.62     (None)      E9F8FCE8-93  00101/00055  unkn  No       Rx:
OPTIONS
192.168.1.62     lluc-softp  7d6f737c67c  00102/00000  unkn  No      
Init: INVITE
11 active SIP channels
-----------------------------------------------------

As you can see, there is a permanent INVITE SIP channel with the user
"lluc-softphone".


Tshi issue occurs sometimes with differents users, but always SJphone
softphones. Any idea?


If you need more help please ask me and I'll try to get it the next time
it occurs.
====================================================================== 

---------------------------------------------------------------------- 
 eliel - 07-04-07 09:49  
---------------------------------------------------------------------- 
Please try to reproduce this bugs in this way:
Call a phone with call-limit=1.
Within this phone call, put onHold the callee, and recover the call from
the onHold state and hangup.
If I am not wrong the called phone (with call-limit=1) will stay in busy
state, even if the call has been hangup.

Example:
phone1 (call-limit=1)
phone2 (client no call-limit)

phone2 --[ call ]-> phone1
phone1 --[ put on hold ]-> phone2
phone1 --[ recover the call (stop on hold) ]-> phone2
phone1 or phone2 --[ hangup ]--

phone1 should stay in BUSY state. (*inuse = 1)

I think this is the problem, I am trying to find a solution. (and I found
this is reproducible in asterisk 1.2 too). 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-04-07 09:49  eliel          Note Added: 0066484                          
======================================================================




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