Asterisk 0.7.0

Changes in 0.7.0 since 0.5.0


VoiceMail and VoiceMailMain were discontinued in favour of VoiceMail2 and VoiceMailMain2
IAX was discontinued in favour of IAX2; to completely remove IAX1 support from Asterisk it is suggested to include "noload => chan_iax.so" in modules.conf. Note that IAX (aka IAX1) is known to cause server crashes after repetitive RELOAD commands.

  • Removed MP3 format and codec
  • Can now load and unload SIP,IAX,IAX2,H323 channels without core
  • Fixed various compiler warnings and clean up source tree
  • Preliminary AES Support
  • Fix SIP REINVITE
  • Outbound SIP registration behind NAT using externip
  • More CLI documentation and clean up
  • Pin numbers on MeetMe
  • Dynamic MeetMe conferences are more consistent with static conferences
  • Added channel variables ${HANGUPCAUSE}, ${SIPDOMAIN}, ${TIMESTAMP}, ${ACCOUNTCODE}
  • ODBC support for logging CDRs
  • Indications for Norway and New Zeland
  • Major redesign of app_voicemail
  • Syslog support
  • Reload logfiles with CLI command 'logger reload' and rotate logs with "logger rotate'
  • Configurable DEBUG, NOTICE, WARNING, ERROR and ast_verbose messages now appear on remote console
  • Properly reaping any zombie processes
  • Added applications SayUnixTime, SetCDRUserField, HasNewVoicemail, ZapScan, Random, ResetCDR, NoCDR
  • Make PRI Hangup Cause available to the dialplan
  • Verify included contexts in extensions.conf
  • Add DESTDIR support for building RPMs and packages
  • Do route lookups on OpenBSD
  • Add support for building on FreeBSD and OS X
  • Add support for PostgreSQL in Voicemail
  • Translate SIP hangup cause to PRI hangup cause where needed
  • Better support for MOH in IAX2
  • Fix SIP problem where channels were not removed on BYE
  • Display codecs by name
  • Remove MySQL and put PGSql instead for licensing reasons
  • Better capability matching in SIP
  • Full IBR4 compliance for chan_zap
  • More flexible CDR handling
  • Distinguish between BUSY and FAILURE on outbound calls
  • Add initial support for SCCP via chan_skinny
  • Better support for Future Group B signaling

Go back to Asterisk status

Changes in 0.7.0 since 0.5.0


VoiceMail and VoiceMailMain were discontinued in favour of VoiceMail2 and VoiceMailMain2
IAX was discontinued in favour of IAX2; to completely remove IAX1 support from Asterisk it is suggested to include "noload => chan_iax.so" in modules.conf. Note that IAX (aka IAX1) is known to cause server crashes after repetitive RELOAD commands.

  • Removed MP3 format and codec
  • Can now load and unload SIP,IAX,IAX2,H323 channels without core
  • Fixed various compiler warnings and clean up source tree
  • Preliminary AES Support
  • Fix SIP REINVITE
  • Outbound SIP registration behind NAT using externip
  • More CLI documentation and clean up
  • Pin numbers on MeetMe
  • Dynamic MeetMe conferences are more consistent with static conferences
  • Added channel variables ${HANGUPCAUSE}, ${SIPDOMAIN}, ${TIMESTAMP}, ${ACCOUNTCODE}
  • ODBC support for logging CDRs
  • Indications for Norway and New Zeland
  • Major redesign of app_voicemail
  • Syslog support
  • Reload logfiles with CLI command 'logger reload' and rotate logs with "logger rotate'
  • Configurable DEBUG, NOTICE, WARNING, ERROR and ast_verbose messages now appear on remote console
  • Properly reaping any zombie processes
  • Added applications SayUnixTime, SetCDRUserField, HasNewVoicemail, ZapScan, Random, ResetCDR, NoCDR
  • Make PRI Hangup Cause available to the dialplan
  • Verify included contexts in extensions.conf
  • Add DESTDIR support for building RPMs and packages
  • Do route lookups on OpenBSD
  • Add support for building on FreeBSD and OS X
  • Add support for PostgreSQL in Voicemail
  • Translate SIP hangup cause to PRI hangup cause where needed
  • Better support for MOH in IAX2
  • Fix SIP problem where channels were not removed on BYE
  • Display codecs by name
  • Remove MySQL and put PGSql instead for licensing reasons
  • Better capability matching in SIP
  • Full IBR4 compliance for chan_zap
  • More flexible CDR handling
  • Distinguish between BUSY and FAILURE on outbound calls
  • Add initial support for SCCP via chan_skinny
  • Better support for Future Group B signaling

Go back to Asterisk status

Created by: JustRumours, Last modification: Wed 28 of Jan, 2004 (11:21 UTC)
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