Asterisk Crash


I am facing a problem when i have use canreinvite=yes than any user transfer call from one sip extension to another sip extension than there is no audio but when i have make canreinvite=no then every thing is working fine expect after same time asterisk crash with segmentation fault. i am using asterisk 1.4.0.


Make sure you do not have any modules installed that do not match the version of the main part of Asterisk.

Upgrade to 1.4.22 or and retry. If you can still reproduce the fault, rebuild with compiler flags set for NOOPTIMIZE and raise a bug report on bugs.digium, providing the information (gdb bt, bt full, and thread apply all bt) specified in the bug reporting guidelines, for crash bugs.

1.4.0 is very old and the only solution for crash bugs in it is to upgrade.