[asterisk-bugs] [Asterisk 0013957]: SIP Channels Hang - Last Message: Rx BYE - Need Destroy: 2
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Dec 16 11:50:15 CST 2008
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=13957
======================================================================
Reported By: geoff2010
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 13957
Category: Channels/chan_sip/General
Reproducibility: random
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.4.21.2
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
======================================================================
Date Submitted: 2008-11-23 18:38 CST
Last Modified: 2008-12-16 11:50 CST
======================================================================
Summary: SIP Channels Hang - Last Message: Rx BYE - Need
Destroy: 2
Description:
I have a bunch of servers running 1.4.21.2. They are all, on occassion,
getting stuck SIP channels. Nothing shows up in "core show channels", but
the lingering culprits continue to show up in "sip show channels" until a
restart of asterisk. The thing they all have in common is that the last
message received was a BYE, and they all have their "Need Destroy" flag set
to 2. Here are some basic outputs, nothing is crashing so I have no core
dumps, it's a production system so I don't have SIP debugging enabled. It
seems to only happen in roughly 1 in 5000 calls (estimate)
atl-asterisk5*CLI> core show channels
Channel Location State Application(Data)
0 active channels
0 active calls
-------------------------------------------------------------------------
-------------------------------------------------------------------------
atl-asterisk5*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Format
Hold Last Message
216.82.XXX.XXX +17066XXX 6b25419936b 00102/00001 0x0 (nothing) No
(d) Rx: BYE
216.82.XXX.XXX +14042XXX 3955ba5d7c2 00102/00001 0x0 (nothing) No
(d) Rx: BYE
2 active SIP channels
-------------------------------------------------------------------------
-------------------------------------------------------------------------
atl-asterisk5*CLI> sip show channel 6b25419936b
atl-asterisk5*CLI>
* SIP Call5*CLI>
Curr. trans. direction: Outgoing
Call-ID: 6b25419936bd5cac4b4dbe6568042787 at blah.com
Owner channel ID: <none>
Our Codec Capability: 260
Non-Codec Capability (DTMF): 1
Their Codec Capability: 256
Joint Codec Capability: 256
Format: 0x0 (nothing)
MaxCallBR: 384 kbps
Theoretical Address: 216.82.XXX.XXX:5060
Received Address: 216.82.XXX.XXX:5060
SIP Transfer mode: open
NAT Support: Always
Audio IP: 67.220.101.185 (local)
Our Tag: as1f608769
Their Tag: VPST506071629460
SIP User agent:
Username: +1706687XXXX
Peername: bw_g729
Original uri: sip:+170668XXXX at 216.82.XXX.XXX
Need Destroy: 2
Last Message: Rx: BYE
Promiscuous Redir: No
Route: N/A
DTMF Mode: rfc2833
SIP Options: (none)
If there is anything else I can provide, please let me know.
Thanks,
Geoff
======================================================================
----------------------------------------------------------------------
(0096459) ivankolev (reporter) - 2008-12-16 11:50
http://bugs.digium.com/view.php?id=13957#c96459
----------------------------------------------------------------------
I have almost exactly the same issue, again on production servers so hard
to provide sip debug output. I'll try to set-up test machine and use Sipp
to generate calls but whether the bug will creep up remains to be seen.
Machine in question takes heavy load of calls, after clean restart and one
day of work I have close to 70 zombie channels.
Here's what sip show channel says:
astappsrv1*CLI> sip show channel
124ad3c3e19f1c3d13e0f45da5925acddff8cdb6 at 205.205.231.22
astappsrv1*CLI>
* SIP Call
Curr. trans. direction: Incoming
Call-ID:
124ad3c3e19f1c3d13e0f45da5925acddff8cdb6 at 205.205.231.22
Owner channel ID: <none>
Our Codec Capability: 262
Non-Codec Capability (DTMF): 1
Their Codec Capability: 2308
Joint Codec Capability: 260
Format: 0x0 (nothing)
MaxCallBR: 384 kbps
Theoretical Address: 205.205.231.22:5060
Received Address: 205.205.231.22:5060
SIP Transfer mode: open
NAT Support: RFC3581
Audio IP: 204.11.120.69 (local)
Our Tag: as76340350
Their Tag: 1c1223205522
SIP User agent: Audiocodes-Sip-Gateway-Mediant
2000/v.4.40.254.482
Peername: pritrunkdomain
Original uri: sip:9056674767 at 205.205.231.22:5060
Caller-ID: 9056674767
Need Destroy: 2
Last Message: Tx: BYE
Promiscuous Redir: No
Route:
sip:9056674767 at 205.205.231.22:5060;nt_end_pt=YM0+~Kn25r6_4t67~QQiE_q25X861aQ5~LKrM2HXoSvi8Q~MA2N03cK1ek_ihlDdkuZXoH1iatPoS10lWUktb-488~Xn7ZQzN~L1tq3U0z3ofW0JQiktq0UJpy~D4190Ius0brUJpvii;nt_server_host=205.205.231.22
DTMF Mode: rfc2833
SIP Options: replaces replace 100rel timer join path
Issue History
Date Modified Username Field Change
======================================================================
2008-12-16 11:50 ivankolev Note Added: 0096459
======================================================================
More information about the asterisk-bugs
mailing list