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Summary:ASTERISK-07982: Asterisk can't dial or register sip devices after a while
Reporter:Francesco Romano (francesco_r)Labels:
Date Opened:2006-10-23 08:16:11Date Closed:2006-10-23 20:39:27
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
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Environment:Attachments:
Description:In two identical machines and similar configurations, asterisk stop to work with sip devices one or two times a day. The core application and the zap side seems continue to work well instead. I must stop and restart asterisk to redial or to reregister sip devices.
Asterisk 1.2.13 is installed in Slackware 10.2 with customised SMP Kernel 2.6.13 and with FreePBX as frontend. All version i tried from 1.2.9 did the same problem. The hardware is Dual Xeon 3GHZ on Intel SE7520AF2 MB with dual PRI Sangoma A102 card.

For example this is few moment before the problem, from the full log:

Oct 23 12:34:40 VERBOSE[4268] logger.c:     -- AGI Script dialparties.agi completed, returning 0
Oct 23 12:34:40 VERBOSE[4268] logger.c:     -- Executing Dial("Local/273@from-internal-67b1,2", "SIP/273||tT") in new stack
Oct 23 12:34:40 DEBUG[4268] chan_sip.c: Setting NAT on RTP to 0
Oct 23 12:34:40 DEBUG[4268] chan_sip.c: Outgoing Call for 273
Oct 23 12:34:40 VERBOSE[4268] logger.c:     -- Called 273
Oct 23 12:34:40 VERBOSE[4268] logger.c:     -- SIP/273-083bfff8 is ringing
Oct 23 12:34:44 VERBOSE[4268] logger.c:     -- SIP/273-083bfff8 answered Local/273@from-internal-67b1,2
Oct 23 12:34:52 DEBUG[4268] channel.c: Didn't get a frame from channel: Local/273@from-internal-67b1,2
Oct 23 12:34:52 DEBUG[4268] channel.c: Bridge stops bridging channels Local/273@from-internal-67b1,2 and SIP/273-083bfff8
Oct 23 12:34:52 DEBUG[4268] chan_sip.c: update_call_counter(273) - decrement call limit counter
Oct 23 12:34:52 DEBUG[4268] app_dial.c: Exiting with DIALSTATUS=ANSWER.

And this is the problem:

Oct 23 12:37:33 VERBOSE[5136] logger.c:     -- AGI Script dialparties.agi completed, returning 0
Oct 23 12:37:33 VERBOSE[5136] logger.c:     -- Executing Dial("Local/220@from-internal-590d,2", "SIP/220||tT") in new stack
Oct 23 12:37:33 DEBUG[5136] chan_sip.c: Setting NAT on RTP to 0
Oct 23 12:37:33 DEBUG[5136] chan_sip.c: Outgoing Call for 220
Oct 23 12:37:33 VERBOSE[5136] logger.c:     -- Called 220
Oct 23 12:37:48 DEBUG[5136] chan_sip.c: update_call_counter(220) - decrement call limit counter
Oct 23 12:37:48 DEBUG[5136] chan_sip.c: Acked pending invite 102
Oct 23 12:37:48 DEBUG[5136] chan_sip.c: Stopping retransmission on '792ae8b736858b1c2b70227e439189cd@192.168.36.15' of Request  102: Match Found
Oct 23 12:37:48 DEBUG[5136] app_dial.c: Exiting with DIALSTATUS=CANCEL.

I have recompile both asterisks with dont-optimize and "DEBUG_THREADS = -DDEBUG_THREADS -DDO_CRASH" options in Makefile but i don't have core dumps or deadlock.
I have other asterisk server installed with Slackware, Freepbx and the same kernel version and works well, but these two are the busiest, sometimes all 60 pri channels are occupied.

I have attached also a piece of the full log, the last successfully call is at 12:34:44 (you can see "build_route: Contact hop: <sip:273@192.168.36.47:5060>")

Thank you
Comments:By: Francesco Romano (francesco_r) 2006-10-23 08:24:14

Sorry for the triple post. Close this and see 8204 please.