On Wed, Apr 9, 2008 at 5:29 PM, Trevor Peirce <[EMAIL PROTECTED]> wrote: > Mindaugas Kezys wrote: > > Hello, > > > > Asterisk 1.4.19 crashes everytime using Realtime and SIP peers > > > Yes I also saw this and had to revert. Calls to the IVR seemed to be > fine, but as soon as two peers call each other it crashes as the call > progresses (never connects). I haven't had a chance to explore any > further and therefore haven't posted a bug either. Perhaps this weekend > if nobody does first.
So far works fine for me. Sample peer setup below. Had one issue with peers where ipaddr was 0 (and hostname used instead), but adding this patch ( http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?r1=113012&r2=113240 ) seems to solve everything. Regards, Atis *************************** 1. row *************************** id: 2 name: 21168 accountcode: NULL amaflags: NULL callgroup: NULL callerid: Atis <21168> canreinvite: no context: default-sip defaultip: NULL dtmfmode: rfc2833 fromuser: NULL fromdomain: NULL fullcontact: sip:[EMAIL PROTECTED]:5061 host: dynamic insecure: NULL language: NULL mailbox: [EMAIL PROTECTED] md5secret: NULL nat: yes deny: NULL permit: NULL mask: NULL pickupgroup: NULL port: 5061 qualify: no restrictcid: NULL rtptimeout: NULL rtpholdtimeout: NULL secret: 21168 type: friend username: 21168 disallow: allow: all musiconhold: NULL regseconds: 1207763735 ipaddr: 192.168.1.123 regexten: cancallforward: yes setvar: call-limit: 4 -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users