[asterisk-users] Asterisk 1.4.19 crash with Realtime using SIP peers

[Atis Lezdins]

On Wed, Apr 9, 2008 at 5:29 PM, Trevor Peirce <tpeirce at digitalcon.ca> 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:21168 at 192.168.1.123:5061
          host: dynamic
      insecure: NULL
      language: NULL
       mailbox: 21168 at device
     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,
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835