No I am not using any realtime config. its text file.. shirley*CLI> core show settings
PBX Core settings ----------------- Version: 1.8.3.2 Build Options: LOADABLE_MODULES Maximum calls: 250 (Current 0) Maximum open file handles: Not set Verbosity: 3 Debug level: 0 Maximum load average: 0.000000 Minimum free memory: 0 MB Startup time: 15:08:59 Last reload time: 15:08:59 System: Linux/2.6.32-24-server built by root on x86_64 2011-03-22 18:38:19 UTC System name: Entity ID: 00:30:48:77:1c:3c Default language: en Language prefix: Enabled User name and group: asterisk/asterisk Executable includes: Disabled Transcode via SLIN: Enabled Internal timing: Enabled Transmit silence during rec: Disabled Generic PLC: Enabled * Subsystems ------------- Manager (AMI): Enabled Web Manager (AMI/HTTP): Disabled Call data records: Enabled Realtime Architecture (ARA): Disabled * Directories ------------- Configuration file: Configuration directory: /etc/asterisk Module directory: /usr/lib/asterisk/modules Spool directory: /var/spool/asterisk Log directory: /var/log/asterisk Run/Sockets directory: /var/run/asterisk PID file: /var/run/asterisk/asterisk.pid VarLib directory: /var/lib/asterisk Data directory: /var/lib/asterisk ASTDB: /var/lib/asterisk/astdb IAX2 Keys directory: /var/lib/asterisk/keys AGI Scripts directory: /var/lib/asterisk/agi-bin > Date: Fri, 8 Apr 2011 11:12:59 -0400 > From: pabelan...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] IAX2/0.0.29.199 > > On 11-04-08 10:48 AM, satish patel wrote: > > > > Where this revers IP comes from ? > > > > == Using SIP RTP CoS mark 5 > > -- Executing [7623@from-sip:1] Macro("SIP/7527-0000006b", > > "stdexten,7623,SIP/7623") in new stack > > -- Executing [s@macro-stdexten:1] ChanIsAvail("SIP/7527-0000006b", > > "SIP/7623&IAX2/7623,20,t") in new stack > > -- Hungup 'IAX2/0.0.29.199:4569-5255' > > -- Executing [s@macro-stdexten:2] NoOp("SIP/7527-0000006b", > > "IAX2/0.0.29.199:4569-5255") in new stack > > -- Executing [s@macro-stdexten:3] NoOp("SIP/7527-0000006b", "0&0") in > > new stack > > -- Auto fallthrough, channel 'SIP/7527-0000006b' status is 'UNKNOWN' > > > Asterisk 1.8? Are you using realtime? Looks to be an issue with > netsock2.c. > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users