[asterisk-bugs] [Asterisk 0018200]: [patch] chan_sip hangs

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Oct 26 10:53:13 CDT 2010


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=18200 
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Reported By:                DennisD
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   18200
Category:                   Channels/chan_sip/General
Reproducibility:            sometimes
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.0 
JIRA:                        
Regression:                 Yes 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-10-23 15:25 CDT
Last Modified:              2010-10-26 10:53 CDT
====================================================================== 
Summary:                    [patch] chan_sip hangs
Description: 
SIP hangs. I'm only using SIP phones (including a Grandstream GXW4008 for
FXS ports).

The phone stop registering and the existing calls quit working
(directmedia=no).

I threw together a quick Perl script to catch it (I've attached it).  I
just check some SIP registrations and that they say we're still registered
and we've gone past our timeout by 120 seconds.  This seems to catch it
every time.

I had written this script while trying to debug this in 1.6<something>,
but 1.6.2.x has worked just fine for many months with the same config
files, so it was fixed there at some point.
====================================================================== 

---------------------------------------------------------------------- 
 (0128404) DennisD (reporter) - 2010-10-26 10:53
 https://issues.asterisk.org/view.php?id=18200#c128404 
---------------------------------------------------------------------- 
This seems to cause the problem within 10-15 minutes almost every time
(with 1.8.0).  Sometimes sooner, sometimes later.

------------

exten => 2229,1,Goto(babymonitor,s,1)

[babymonitor]
exten => s,1,NoOp()
same => n,Set(DENOISE(rx)=on)
same => n,Set(DENOISE(tx)=on)
same => n,Set(AGC(rx)=${AGCLEVEL})
same => n,Set(AGC(tx)=${AGCLEVEL})
same => n,Set(GROUP()=2229)
same => n,NoOp(GROUP_COUNT(${exten}): ${GROUP_COUNT(2229)})
same => n,GotoIf($[ ${DEVICE_STATE(SIP/2024)} = INUSE ]?inuse)
;same => n,SIPAddHeader(Call-Info: sip:\;answer-after=0)    ; This isn't
being sent by Originate.  :(
same => n,Originate(SIP/2024,app,ConfBridge,2024,1q)
same => n,NoOp(ORIGINATE_STATUS: ${ORIGINATE_STATUS})
same => n,GotoIf($[ ${ORIGINATE_STATUS} = SUCCESS ]?inuse)
same => n,AGI(flite.pl,"ORIGINATE STATUS: ${ORIGINATE_STATUS}")
same => n(inuse),Answer()
same => n,PlayTones(300)
same => n,Wait(0.15)
same => n,StopPlayTones()
same => n,ConfBridge(2024,1qMm)

exten => h,1,NoOp(GROUP_COUNT(${exten}): ${GROUP_COUNT(2229)})
same => n,ExecIf($[ ${GROUP_COUNT(2229)} = 1 ]?SoftHangup(SIP/2024,a)

------------

Yes, there is an old GXP-2000 under the crib set to auto answer and the
volume (ringing and speakerphone) set to the lowest possible (you can't
hear it).  This works exceptionally well with 1.6.2.14-rc1 (and previous
versions).

With 1.6.2, I do get messages like:

[2010-10-25 17:01:42.570] WARNING[22580] chan_sip.c: Maximum retries
exceeded on transmission a51c9bb2d05d402e at 192.168.x.x for seqno 182
(Critical Request) -- See doc/sip-retransmit.txt.
[2010-10-25 17:01:42.570] WARNING[22580] chan_sip.c: Hanging up call
a51c9bb2d05d402e at 192.168.x.x - no reply to our critical packet (see
doc/sip-retransmit.txt).

(in 1.6.2.x) this will drop the call, but it happens once every couple of
days and sometimes the call is forgotten about and left up for 6 or more
hours.  If I set a phone to use G722, I'll get that message and drop the
call within about 10-15 minutes.  Everything else is using ulaw. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-10-26 10:53 DennisD        Note Added: 0128404                          
======================================================================




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