[asterisk-bugs] [Asterisk 0016185]: Crash of Outgoing Call
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Nov 10 23:44:03 CST 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16185
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Reported By: alex70
Assigned To: mnicholson
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Project: Asterisk
Issue ID: 16185
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: assigned
Asterisk Version: SVN
JIRA: SWP-365
Regression: Yes
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.2
SVN Revision (number only!): 227947
Request Review:
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Date Submitted: 2009-11-05 06:26 CST
Last Modified: 2009-11-10 23:44 CST
======================================================================
Summary: Crash of Outgoing Call
Description:
Everyday our asterisk system updates itself with the most recent svn
branch.
Last night asterisk update to revision r227947 and every outbound call
falls as soon as the called pick up the phone.
Our suspect is there is something wrong with the patch uploaded by
mnicholson to resolve issue https://issues.asterisk.org/view.php?id=16005 with:
R227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov 2009) | 17
lines - "This patch modifies the Dial application to monitor the calling
channel for hangups while playing back announcements"
Please see also attached verbosity file
======================================================================
----------------------------------------------------------------------
(0113640) xblurone (reporter) - 2009-11-10 23:44
https://issues.asterisk.org/view.php?id=16185#c113640
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I am having exactly the same issue... here an excerpt from the full log
file... As soon as the call is answered, asterisk hangs-up the call...
unfortunately I can't downgrade to 1.6.1.9 as asterisk then dies at random
every day after a few hours.
Sip client -> Dahdi/PRI PSTN works fine
Sip client -> Sip provider PSTN hangs up after answer. (90% of the time)
[2009-11-11 13:24:08] DEBUG[5001] pbx.c: Launching 'Dial'
[2009-11-11 13:24:08] VERBOSE[5001] pbx.c: [2009-11-11 13:24:08] --
Executing [98765432 at aglow:1] Dial("SIP/63115000-00000034",
"SIP/https://issues.asterisk.org/view.php?id=1146598765432@xxx.yyy.zzz.177,40")
in new stack
[2009-11-11 13:24:08] DEBUG[5001] chan_sip.c: Asked to create a SIP
channel with formats: 0x1 (g723)
[2009-11-11 13:24:08] VERBOSE[5001] netsock.c: [2009-11-11 13:24:08] ==
Using SIP RTP TOS bits 184
[2009-11-11 13:24:08] VERBOSE[5001] netsock.c: [2009-11-11 13:24:08] ==
Using SIP RTP CoS mark 5
[2009-11-11 13:24:08] VERBOSE[5001] netsock.c: [2009-11-11 13:24:08] ==
Using SIP VRTP TOS bits 136
[2009-11-11 13:24:08] VERBOSE[5001] netsock.c: [2009-11-11 13:24:08] ==
Using SIP VRTP CoS mark 6
[2009-11-11 13:24:08] VERBOSE[5001] netsock.c: [2009-11-11 13:24:08] ==
Using UDPTL TOS bits 184
[2009-11-11 13:24:08] VERBOSE[5001] netsock.c: [2009-11-11 13:24:08] ==
Using UDPTL CoS mark 5
[2009-11-11 13:24:08] DEBUG[5001] chan_sip.c: Allocating new SIP dialog
for 75ba8b8570a885191c51f28675e8356f at 202.42.66.17 - INVITE (With RTP)
[2009-11-11 13:24:08] DEBUG[5001] chan_sip.c: Setting NAT on RTP to Off
[2009-11-11 13:24:08] DEBUG[5001] chan_sip.c: Setting NAT on UDPTL to Off
[2009-11-11 13:24:08] DEBUG[5001] channel.c: Not copying variable
DIALEDTIME.
[2009-11-11 13:24:08] DEBUG[5001] channel.c: Not copying variable
ANSWEREDTIME.
[2009-11-11 13:24:08] DEBUG[5001] channel.c: Not copying variable
DIALEDPEERNAME.
[2009-11-11 13:24:08] DEBUG[5001] channel.c: Not copying variable
DIALEDPEERNUMBER.
[2009-11-11 13:24:08] DEBUG[5001] channel.c: Not copying variable
DIALSTATUS.
[2009-11-11 13:24:08] DEBUG[5001] channel.c: Not copying variable
SIPCALLID.
[2009-11-11 13:24:08] DEBUG[5001] channel.c: Not copying variable
SIPDOMAIN.
[2009-11-11 13:24:08] DEBUG[5001] channel.c: Not copying variable SIPURI.
[2009-11-11 13:24:08] DEBUG[5001] chan_sip.c: Outgoing Call for
https://issues.asterisk.org/view.php?id=1146598765432
[2009-11-11 13:24:08] DEBUG[5001] chan_sip.c: ** Our capability: 0x100
(g729) Video flag: False Text flag: False
[2009-11-11 13:24:08] DEBUG[5001] chan_sip.c: ** Our prefcodec: 0x1 (g723)
[2009-11-11 13:24:08] DEBUG[5001] chan_sip.c: Initializing initreq for
method INVITE - callid 6dfc0e524a7a2805201a3c470c008a98 at xxx.yyy.zzz.134
[2009-11-11 13:24:08] DEBUG[5001] chan_sip.c: Trying to put 'INVITE sip:'
onto UDP socket destined for xxx.yyy.zzz.177:5060
[2009-11-11 13:24:08] VERBOSE[5001] app_dial.c: [2009-11-11 13:24:08]
-- Called https://issues.asterisk.org/view.php?id=1146598765432@xxx.yyy.zzz.177
[2009-11-11 13:24:14] VERBOSE[5001] app_dial.c: [2009-11-11 13:24:14]
-- SIP/xxx.yyy.zzz.177-00000035 is making progress passing it to
SIP/63115000-00000034
[2009-11-11 13:24:14] DEBUG[5001] chan_sip.c: ** Our capability: 0x1
(g723) Video flag: True Text flag: True
[2009-11-11 13:24:14] DEBUG[5001] chan_sip.c: ** Our prefcodec: 0x0
(nothing)
[2009-11-11 13:24:14] DEBUG[5001] chan_sip.c: Trying to put 'SIP/2.0 183'
onto UDP socket destined for xxx.yyy.zzz.168:5060
[2009-11-11 13:24:14] DEBUG[5001] rtp.c: Ooh, format changed from unknown
to g729
[2009-11-11 13:24:14] DEBUG[5001] rtp.c: Created smoother: format: 256 ms:
20 len: 20
[2009-11-11 13:24:14] DEBUG[5001] rtp.c: Ooh, format changed from unknown
to g723
[2009-11-11 13:24:14] DEBUG[5001] rtp.c: Got RTCP report of 112 bytes
[2009-11-11 13:24:17] DEBUG[5001] rtp.c: Got RTCP report of 112 bytes
[2009-11-11 13:24:17] VERBOSE[5001] app_dial.c: [2009-11-11 13:24:17]
-- SIP/xxx.yyy.zzz.177-00000035 answered SIP/63115000-00000034
[2009-11-11 13:24:17] DEBUG[5001] chan_sip.c: SIP answering channel:
SIP/63115000-00000034
[2009-11-11 13:24:17] DEBUG[5001] chan_sip.c: ** Our capability: 0x1
(g723) Video flag: True Text flag: True
[2009-11-11 13:24:17] DEBUG[5001] chan_sip.c: ** Our prefcodec: 0x0
(nothing)
[2009-11-11 13:24:17] DEBUG[5001] chan_sip.c: Trying to put 'SIP/2.0 200'
onto UDP socket destined for xxx.yyy.zzz.168:5060
[2009-11-11 13:24:22] DEBUG[5001] channel.c: Didn't get a frame from
channel: SIP/63115000-00000034
[2009-11-11 13:24:22] DEBUG[5001] channel.c: Bridge stops bridging
channels SIP/63115000-00000034 and SIP/xxx.yyy.zzz.177-00000035
[2009-11-11 13:24:22] DEBUG[5001] channel.c: Hanging up channel
'SIP/xxx.yyy.zzz.177-00000035'
[2009-11-11 13:24:22] DEBUG[5001] chan_sip.c: Hangup call
SIP/xxx.yyy.zzz.177-00000035, SIP callid
6dfc0e524a7a2805201a3c470c008a98 at xxx.yyy.zzz.134
[2009-11-11 13:24:22] DEBUG[5001] chan_sip.c: Trying to put 'BYE sip:116'
onto UDP socket destined for xxx.yyy.zzz.177:5060
[2009-11-11 13:24:22] DEBUG[5001] rtp.c: Channel '<unspecified>' has no
RTP, not doing anything
[2009-11-11 13:24:22] DEBUG[5001] app_dial.c: Exiting with
DIALSTATUS=ANSWER.
[2009-11-11 13:24:22] DEBUG[5001] pbx.c: Spawn extension
(aglow,98765432,1) exited non-zero on 'SIP/63115000-00000034'
[2009-11-11 13:24:22] VERBOSE[5001] pbx.c: [2009-11-11 13:24:22] ==
Spawn extension (aglow, 98765432, 1) exited non-zero on
'SIP/63115000-00000034'
[2009-11-11 13:24:22] DEBUG[5001] channel.c: Soft-Hanging up channel
'SIP/63115000-00000034'
[2009-11-11 13:24:22] DEBUG[5001] channel.c: Hanging up channel
'SIP/63115000-00000034'
[2009-11-11 13:24:22] DEBUG[5001] chan_sip.c: Hangup call
SIP/63115000-00000034, SIP callid e3617b44-8d4fcd2b at xxx.yyy.zzz.168
Issue History
Date Modified Username Field Change
======================================================================
2009-11-10 23:44 xblurone Note Added: 0113640
======================================================================
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