[asterisk-bugs] [JIRA] (ASTERISK-22903) SIP Hangs, Asterisk becomes unresponsive
Craig Arno (JIRA)
noreply at issues.asterisk.org
Mon Nov 25 17:14:03 CST 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-22903?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=212251#comment-212251 ]
Craig Arno commented on ASTERISK-22903:
---------------------------------------
Here are the two basic peer configurations which have mostly worked as is since Asterisk 1.0 (with a few syntax changes). And as I said, kill Asterisk process, then rerun and everything works again with no configuration changes.
{code}
[LINE2]
type=peer
host=dynamic
localnet=192.168.90.0/24
secret=mySecret
callerid="Craig" <52>
dtmfmode=rfc2833
dtmf=rfc2833
relaxdtmf=yes ;YES=Interpret DTMF tones during a call (for "outside" RFC2833 channel)
toneduration=150
insecure=port,invite
canreinvite=no ;Asterisk must stay in the middle because SPA2100 is behind NAT
nat=no ;At home, so no NAT between SPA2100 and Asterisk
register => 4255551212 at sip.broadvoice.com:myBVsecret:4255551212 at sip.broadvoice.com/4255551212
[sip.broadvoice.com]
;Following is new with Asterisk 1.8
defaultuser=4255551212
;End of new 1.8 stuff
type=peer
user=phone
qualify=yes
;NAT=yes - only if Asterisk is behind a NAT (myDomain.com is not)
nat=no
;Disable canreinvite if SIP device is behind a NAT, BV doesn't allow reinvite
canreinvite=no
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=4255551212
secret=0v01hwstyk
username=4255551212
insecure=port,invite ;"port,invite" works. "very" is another setting.
context=from-broadvoice
authname=4255551212
{code}
> SIP Hangs, Asterisk becomes unresponsive
> ----------------------------------------
>
> Key: ASTERISK-22903
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-22903
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Affects Versions: 11.6.0
> Environment: OpenSUSE 12.2 x64 (uname -a):
> Linux pluto 3.4.63-2.44-desktop #1 SMP PREEMPT Wed Oct 2 11:18:32 UTC 2013 (d91a619) x86_64 x86_64 x86_64 GNU/Linux
> AMD FX-8120 processor.
> Reporter: Craig Arno
> Assignee: Craig Arno
>
> I'm here looking for a solution to a SIP/Asterisk hang issue which showed first in Asterisk version 11.6.0. I rolled back to 11.5.1 which works fine on my server.
> When I figure out my Asterisk 11.6.0 installation isn't accepting calls, I execute "core stop now", and am told Asterisk is stopping, and everything looks/behaves normal. There is no running Asterisk process. I then restart Asterisk from the command prompt, no options and it looks like it boots normally, except Asterisk will then not respond to commands, like "core stop now". Asterisk also won't quit with a normal "kill" command. The only way to stop Asterisk 11.6.0 is to "kill -9".
> At this point, starting Asterisk 11.6.0 yet another time and it will execute normally for about another day before sliding silently into this "dead" behavior again.
> My call volume is very low, only about 1-5/day. All my Sipura (2000, 2100, 3000, etc) devices are connected using SIP. My hardware is AMD FX-8120, ASUS Sabertooth 990FX, 8GB RAM, 9 TB HD... running under OpenSUSE 12.2 x64; Linux pluto 3.4.63-2.44-desktop #1 SMP PREEMPT Wed Oct 2 11:18:32 UTC 2013 (d91a619) x86_64. Asterisk 11.6.0 is compiled with optimization off and thread debugging on. There are no resource issues and all parts are running to factory specifications. This Asterisk is exposed to the Internet for 2 SIP users and incoming/outgoing calls. This is a stable system (not a new installation).
> I've been silent hoping for another release soon. This looks like a difficult problem to solve, so if I can perform some actions to help without risking an undetectable loss of phone service, please let me know how. 11.5.1 is continuing to run well for me.
> After reviewing 11.6.0 issues, ASTERISK-22854 looks the closest to what I may be experiencing.
> I tried the forums http://forums.asterisk.org/viewtopic.php?f=1&t=88593
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