Summary: | ASTERISK-08120: "Dropping voice to exceptionally long" | ||
Reporter: | Rene Cunningham (rene) | Labels: | |
Date Opened: | 2006-11-12 20:53:26.000-0600 | Date Closed: | 2006-11-20 22:01:54.000-0600 |
Priority: | Critical | Regression? | No |
Status: | Closed/Complete | Components: | Core/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | Hi, Asterisk queues become unresponsive when certain callers enter our queues. Im unable to reproduce this error. The problem occurs sporadically, though it happened 4 times today. Asterisk doesnt respond to 'show queues', 'show channels' or 'show agents'. Also no callers can reach our queues and get 'dead air' when selecting an option from the IVR. Log below (IP address and usernames replaced with X) Nov 13 11:08:39 VERBOSE[11143] logger.c: -- Executing Ringing("IAX2/XXXXXX-2", "") in new stack Nov 13 11:08:39 VERBOSE[11143] logger.c: -- Executing Wait("IAX2/XXXXXX-2", "2") in new stack Nov 13 11:08:41 VERBOSE[11143] logger.c: -- Executing Queue("IAX2/XXXXXX-2", "general-inq|t|||90") in new stack Nov 13 11:08:41 VERBOSE[11143] logger.c: -- Started music on hold, class 'default', on channel 'IAX2/XXXXXX-2' Nov 13 11:08:41 DEBUG[11143] channel.c: Scheduling timer at 160 sample intervals Nov 13 11:08:42 VERBOSE[4881] logger.c: -- Remote UNIX connection Nov 13 11:08:42 VERBOSE[11159] logger.c: -- Remote UNIX connection disconnected Nov 13 11:08:43 DEBUG[4897] chan_sip.c: Setting NAT on RTP to 524288 Nov 13 11:08:43 DEBUG[4897] chan_sip.c: Stopping retransmission on 'b7889b8a-6bccdd84@XXX.XX.XX.XXX' of Response 101: Match Found Nov 13 11:08:43 DEBUG[4897] chan_sip.c: Setting NAT on RTP to 524288 Nov 13 11:08:43 DEBUG[4897] chan_sip.c: Checking SIP call limits for device xxxxxx_desk Nov 13 11:08:43 DEBUG[4897] chan_sip.c: build_route: Contact hop: "XXXXX" <sip:xxxxx@XXX.XX.XX.XXX:5060> Nov 13 11:08:43 DEBUG[4883] channel.c: Avoiding initial deadlock for 'SIP/xxxxxx_desk-0a0bae98' Nov 13 11:08:43 VERBOSE[11162] logger.c: -- Executing Dial("SIP/xxxxx_desk-0a0bae98", "IAX2/xxxxx/XXXXXXXX") in new stack Nov 13 11:08:43 VERBOSE[11162] logger.c: -- Called xxxxx/XXXXXXXX Nov 13 11:08:43 VERBOSE[4893] logger.c: -- Call accepted by XXX.XX.XX.XXX (format alaw) Nov 13 11:08:43 VERBOSE[4893] logger.c: -- Format for call is alaw Nov 13 11:08:43 DEBUG[4893] channel.c: Dropping voice to exceptionally long queue on IAX2/XXXXXX-2 Nov 13 11:08:43 DEBUG[4893] channel.c: Dropping voice to exceptionally long queue on IAX2/XXXXXX-2 Nov 13 11:08:43 DEBUG[4893] channel.c: Dropping voice to exceptionally long queue on IAX2/XXXXXX-2 Nov 13 11:08:43 DEBUG[4893] channel.c: Dropping voice to exceptionally long queue on IAX2/XXXXXX-2 Ive seen a few issues also reporting the same symptoms though no real solutions that may help me out * http://bugs.digium.com/bug_view_advanced_page.php?bug_id=4767&history=1 * http://bugs.digium.com/view.php?id=2896 * http://lists.digium.com/pipermail/asterisk-users/2004-March/034210.html * http://lists.digium.com/pipermail/asterisk-dev/2004-January/003028.html * http://threebit.net/mail-archive/asterisk-dev/msg00042.html ****** ADDITIONAL INFORMATION ****** * Our callers are using IAX and our agents use SIP. * Our calls use ILBC as the codec. Our agents use alaw. OS -- Fedora Core 5 Software -------- kernel 2.6.17 asterisk 1.2.10 zaptel kernel module 1.2.7 | ||
Comments: | By: Serge Vecher (serge-v) 2006-11-17 09:15:57.000-0600 there was some work done on the locking code for app_queue between 1.2.10 and 1.2.13. Can you please test 1.2.13 and see if the problem is fixed there? By: Rene Cunningham (rene) 2006-11-17 18:59:32.000-0600 Ive upgraded to 1.2.13 and have not experienced this problem yet. Thanks! By: Serge Vecher (serge-v) 2006-11-20 22:01:53.000-0600 well, I'll close the issue then. If the problem does resurface, please reopen the report. And remember bug rule #1: "always test the latest release before posting a bug". Thanks, |