I work with various fixes in the Asterisk source tree... cross-compilation to new platforms, adding new features (channels, resources, etc), and adding new configuration samples that do useful things:
https://issues.asterisk.org/view.php?id=16090 https://issues.asterisk.org/view.php?id=15858 https://issues.asterisk.org/view.php?id=15857 https://issues.asterisk.org/view.php?id=12293 https://issues.asterisk.org/view.php?id=11969 https://issues.asterisk.org/view.php?id=11487 and I'd like to do more: * simplify SLA (shared line appearance) through macros, etc. * add E.164 support * add Freenum/ISN support * add diversion support (P-Asserted-Identity, etc) to SIP * add find-me/follow-me via dialplan rules * add telephone auto-configuration hooks etc. if anyone is especially good at troubleshooting dialplan and sip (i.e. extensions.conf and sip.conf) issues, that would be really useful. If you can help me polish some examples of how to do these, I'll file "documentation enhancement fixes" in mantis (the Asterisk bug tracker) and get the changes integrated into the source tree, as I did above. Thanks! -Philip -- _____________________________________________________________________ -- 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
