Summary: | ASTERISK-24603: Channels are stucked in Ring State on Dial Application | ||
Reporter: | ankit (ankitgupta.nt) | Labels: | |
Date Opened: | 2014-12-10 06:57:01.000-0600 | Date Closed: | 2014-12-23 13:51:45.000-0600 |
Priority: | Major | Regression? | |
Status: | Closed/Complete | Components: | |
Versions: | Frequency of Occurrence | Frequent | |
Related Issues: | |||
Environment: | CentOS release 6.5 x86_64 kernel version 2.6.32-431.el6.x86_64 Using only VOIP on SIP | Attachments: | |
Description: | I am facing this issues on frequently, Channels are got stuck and not release due to calling hamper and all are SIP extension showing 408 timeout after restart asterisk all are working fine. | ||
Comments: | By: Matt Jordan (mjordan) 2014-12-10 07:00:08.110-0600 Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. You may find it helpful to read the Asterisk Issue Guidelines http://www.asterisk.org/developers/bug-guidelines. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need: 1. the specific steps or actions you took that caused you to encounter the problem, 2. the behavior you expected, and 3. the behavior you actually encountered (in as much detail as possible). This likely includes output from the console with debug level logging, a SIP trace (if this is SIP related), and configuration information such as dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf). Thanks! By: ankit (ankitgupta.nt) 2014-12-23 02:44:11.610-0600 Sorry to late reply I am sending you the locking and sip.conf kindly see the Below Log and reply me. asterisk -V Asterisk 1.6.2.24 [general] context=default port=5060 bindaddr=0.0.0.0 ;allowguest=no ;alwaysauthreject=yes srvlookup=yes rtptimeout=3600 rtpholdtimeout=3600 externip=192.168.0.2 localnet=10.50.130.14/255.255.255.252 defaultexpiry=3600 [101] type=friend callerid="101" defaultuser=101 secret=456cms insecure=port,invite host=dynamic nat=yes dial=SIP/101 regcontext=from-internal regexten=101 dtmfmode=rfc2833 canreinvite=yes qualify=yes context=from-internal mailbox=101@default callwaiting=no asterisk -rx "core show locks" ======================================================================= === Currently Held Locks ============================================== ======================================================================= === === <pending> <lock#> (<file>): <lock type> <line num> <function> <lock name> <lock addr> (times locked) === === Thread ID: 0x7fa1e0094700 (do_monitor started at [23316] chan_sip.c restart_monitor()) === ---> Lock #0 (astobj2.c): MUTEX 164 ao2_lock &p->priv_data.lock 0x20d77d0 (1) /usr/sbin/asterisk(ast_bt_get_addresses+0xe) [0x4fc6de] /usr/sbin/asterisk() [0x4430c7] /usr/sbin/asterisk(_ao2_callback+0x31) [0x4434f1] /usr/lib/asterisk/modules/chan_sip.so(+0x79e0a) [0x7fa1ebac4e0a] /usr/sbin/asterisk() [0x596cdd] /lib64/libpthread.so.0() [0x3231a079d1] /lib64/libc.so.6(clone+0x6d) [0x32316e8b6d] === ---> Tried and failed to get Lock #1 (chan_sip.c): MUTEX 23196 check_rtp_timeout &dialog->owner->lock_dont_use 0x7fa1b4017dd0 (0) /usr/sbin/asterisk(ast_bt_get_addresses+0xe) [0x4fc6de] /usr/lib/asterisk/modules/chan_sip.so(+0x15d9b) [0x7fa1eba60d9b] /usr/lib/asterisk/modules/chan_sip.so(+0x35caf) [0x7fa1eba80caf] /usr/sbin/asterisk() [0x442828] /usr/sbin/asterisk(_ao2_callback+0x31) [0x4434f1] /usr/lib/asterisk/modules/chan_sip.so(+0x79e0a) [0x7fa1ebac4e0a] /usr/sbin/asterisk() [0x596cdd] /lib64/libpthread.so.0() [0x3231a079d1] /lib64/libc.so.6(clone+0x6d) [0x32316e8b6d] === ------------------------------------------------------------------- === === Thread ID: 0x7fa1bb83f700 (pbx_thread started at [ 4630] pbx.c ast_pbx_start()) === ---> Lock #0 (channel.c): MUTEX 2769 __ast_read &chan->lock_dont_use 0x7fa1b4017dd0 (1) /usr/sbin/asterisk(ast_bt_get_addresses+0xe) [0x4fc6de] /usr/sbin/asterisk() [0x478aea] /usr/lib/asterisk/modules/app_chanspy.so(+0x4182) [0x7fa1ee9af182] /usr/lib/asterisk/modules/app_chanspy.so(+0x9581) [0x7fa1ee9b4581] /usr/lib/asterisk/modules/app_chanspy.so(+0xb406) [0x7fa1ee9b6406] /usr/sbin/asterisk(pbx_exec+0x110) [0x5212a0] /usr/sbin/asterisk() [0x532f05] /usr/sbin/asterisk() [0x537137] /usr/sbin/asterisk() [0x538499] /usr/sbin/asterisk() [0x596cdd] /lib64/libpthread.so.0() [0x3231a079d1] /lib64/libc.so.6(clone+0x6d) [0x32316e8b6d] === ------------------------------------------------------------------- === ======================================================================= Regards Ankit Gupta By: Matt Jordan (mjordan) 2014-12-23 13:52:12.168-0600 Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.4, 1.6.x, and 1.8 branches has ended. For continued maintenance support please move to the 11 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions. After testing with Asterisk 11, if you find this problem has not been resolved, please open a new issue against Asterisk 11. |