|Summary:||ASTERISK-08814: svn last check report|
|Reporter:||Badalian Vyacheslav (slavon)||Labels:|
|Date Opened:||2007-02-15 02:35:44.000-0600||Date Closed:||2007-03-20 09:22:26|
|Environment:||Attachments:||( 0) inuse.txt|
1. "restart gracefully" and "restart when convenient" command freeze console to 1 minute.... when console brick down... when asterisk -r say that asterisk not loaded.... "ps ax | grep asterisk" say that asterisk not loaded =(
2. sip show inuse some time say "-1" on some peer +) funny =)
3. before any call "Call on SIP/cisco3600-trunk-08258c08 left from hold". its normal?
This bugreport as "maybe you don't know about this interesting things" +)
****** ADDITIONAL INFORMATION ******
asterisk asterisk # svn info
Repository Root: http://svn.digium.com/svn/asterisk
Repository UUID: 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Node Kind: directory
Last Changed Author: file
Last Changed Rev: 54552
Last Changed Date: 2007-02-15 05:11:34 +0300 (Thu, 15 Feb 2007)
|Comments:||By: Badalian Vyacheslav (slavon) 2007-02-15 02:38:47.000-0600|
asterisk*CLI> core show version
Asterisk SVN-trunk-r54372 built by root @ asterisk on a i686 running Linux on 2007-02-14 17:01:17 UTC
By: Olle Johansson (oej) 2007-02-15 06:15:42.000-0600
Please don't mix several issues in one bug report. Since this is filed in the SIP category, I'll take care of those.
Please try with canreinvite=no and see what happens with "inuse". thanks.
By: Joshua C. Colp (jcolp) 2007-02-15 08:54:11.000-0600
The console purposely doesn't accept any input until all calls are done for your first statement. Are there still calls up?
By: Badalian Vyacheslav (slavon) 2007-02-16 00:34:17.000-0600
oej - all my peers (friends) what have call-limit have canreinvite=no
file - what i do:
1. core show channels = have 3 active channels
2. restart when convenient or restart when convenient = console freeze... i can't put any charters...
3. open another console... do asterisk -r.... normal login to asterisk... monitoring 2 console.... after 10-15 sec all console logoff from asterisk... ps ax | grep asterisk say what it not start....
i put this bug report all in one because i don't know what of this really bag...
By: Badalian Vyacheslav (slavon) 2007-02-16 00:40:29.000-0600
See inuse.txt... u can see -1 in inUse and also i only have 3 real active channels but very many InUse=1 in show sip inuse...
it alse refer to bug ID "0007744: Call limits"
By: Serge Vecher (serge-v) 2007-03-05 14:27:08.000-0600
slavon, can we please get a full sip debug from 1.4.1?
By: Badalian Vyacheslav (slavon) 2007-03-20 08:35:38
i turn off call-limits. Problems from clients =(
By: Serge Vecher (serge-v) 2007-03-20 09:22:15
ok, one user reported successful results with 1.4.1. If you decide to enable call-limits again and it does not work, please reopen the bug.