Summary:ASTERISK-11599: SIP channel isn't closed when using TLS transport
Reporter:pj (pj)Labels:
Date Opened:2008-03-07 15:40:10.000-0600Date Closed:2008-08-13 16:00:29
Versions:Frequency of
Environment:Attachments:( 0) 20080702__issue12170_clear_pendinginvite.diff
( 1) cli.txt
( 2) sip-debug.txt
Description:when H323 endpoint calls SIP, call is established and then h323 hangs up, asterisk doesn't send sip BYE to sip endpoint and thus channel remains open until RTP times out.
when I tried to setup call in oposite direction, ie. sip endpoint calls h323, call is established, then h323 hangs up, sip BYE is send and channel is correctly closed.
I'm not observing this issue, when using udp as sip signaling transport.


attached sip debug shows two cases, udp and tls transport in use,
in both cases, call was established in direction when H323 calls SIP, call was answered and then H323 hangs up.

Comments:By: Olle Johansson (oej) 2008-07-02 04:44:11

Housekeeping: Any updates? Anyone looking into this? Does the problem still exist?

By: Brett Bryant (bbryant) 2008-07-02 17:53:30

pj, would you mind testing the patch I just uploaded to see if it fixes the issue for you?

By: pj (pj) 2008-07-10 08:08:34

sorry for delay, but I must wait until issue 0012494 will be resolved, I haven't pure testing asterisk and issue 0012494 breaks even basic call functionality for my users. thanks for patience.

By: Paul Belanger (pabelanger) 2008-07-23 16:11:31

Is there anything holding this up for commit?  We have been running this patch for +2 weeks with no problems against issue 0012700.

Thanks again,

By: pj (pj) 2008-07-24 16:09:30

I can't confirm if it working, because asterisk locks, when try to use tls transport between two asterisks
Asterisk SVN-trunk-r133448 +  20080702__issue12170_clear_pendinginvite.diff

=== Thread ID: -1215231088 (do_monitor           started at [19552] chan_sip.c restart_monitor())
=== ---> Lock #0 (chan_sip.c): MUTEX 19522 do_monitor &monlock 0xb79f6280 (1)
       asterisk(ast_bt_get_addresses+0x19) [0x80f2973]
       /usr/lib/asterisk/modules/chan_sip.so [0xb79801d7]
       /usr/lib/asterisk/modules/chan_sip.so [0xb79d06fc]
       asterisk [0x81670b2]
       /lib/i686/libpthread.so.0 [0xb7c7c315]
       /lib/i686/libc.so.6(clone+0x5e) [0xb7d6adde]

By: pj (pj) 2008-07-31 12:46:32

Bret, small notice: this issue was originaly found only when using TLS as sip transport, I don't think, that can be related to ASTERISK-1263700

By: Brett Bryant (bbryant) 2008-07-31 12:51:21

pj, a patch for this issue didn't apply to just TLS, but it did solve the problem and the one in ASTERISK-12063 as well.

By: pj (pj) 2008-07-31 13:10:09

do you have some idea, about lock from my previous test?
Should I try again and provide some more debug?

By: Brett Bryant (bbryant) 2008-07-31 13:23:21

pj, I didn't had a chance to look at it. Have you made sure that the problem happens reliably only with the patch? It seems quite odd that this patch would cause that issue.

By: pj (pj) 2008-07-31 14:01:03

I don't know, if lock was caused by your patch.
But tls seems to be somehow buggy, I have more issues with tls, eg. ASTERISK-1303117
so, If you think, that your patch solves issue from this bugreport (not sending BYE to end call), you can try to commit and close this bugreport.
When tls will better usable later, I will try again and if 'BYE issue' will appear again, I will reopen this bugreport.

By: pj (pj) 2008-08-02 04:18:15

sorry, I attach cli.txt to wrong bugreport, it's related to ASTERISK-1303117
please delete cli.txt from here.

By: Digium Subversion (svnbot) 2008-08-13 16:00:25

Repository: asterisk
Revision: 137532

U   trunk/channels/chan_sip.c

r137532 | qwell | 2008-08-13 16:00:24 -0500 (Wed, 13 Aug 2008) | 8 lines

Correctly end locally ended calls.

(closes issue ASTERISK-11599)
Reported by: pj
     20080702__issue12170_clear_pendinginvite.diff uploaded by bbryant (license 36)
Tested by: bbryant, pabelanger