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pkgsrc/comms/asterisk/PLIST.common_end

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Update asterisk to 1.2.14.
2006-12-20 12:34:55 +01:00
@comment $NetBSD: PLIST.common_end,v 1.4 2006/12/20 11:34:55 mjl Exp $
Changes 1.0.9: -- fix bug in callerid matching in the dialplan that was introduced in 1.0.8 Changes 1.0.8: -- chan_zap -- Asterisk will now also look in the regular context for the fax extension while executing a macro. Previously, for this to work, the fax extension would have to be included in the macro definition. -- On some systems, ALERTING will be sent after PROCEEDING, so code has been added to account for this case. -- If no extension is specified on an overlap call, the 's' extension will be used. -- chan_sip -- We no longer send a "to" tag on "100 Trying" messages, as it is inappropriate to do so. -- We now respond correctly to an invite for T.38 with a "488 Not acceptable here" -- We now discard saved tags on 401/407 responses in case the provider we're talking to tries to pull a dirty trick on us and change it. -- rtptimeout options will now be correctly set on a peer basis rather than only global -- chan_mgcp -- Fixed setting of accountcode -- Fixed where *67 to block callerid only worked for first call -- chan_agent -- We now will not pass audio until the agent has acked the call if the configuration is set up for the agent to do so. -- chan_alsa -- Fixed problems with the unloading of this module -- res_agi -- A fix has been added to prevent calls from being hung up when more than one call is executing an AGI script calling the GET DATA command. -- AGI scripts will now continue to run even if a file was not found with the GET DATA command. -- When calling SAY NUMBER with a number like 09, we will now say "nine" instead of "zero" -- app_dial -- There was a problem where text frames would not be forwarded before the channel has been answered. -- app_disa -- Fixed the timeout used when no password is set -- app_queue -- Distinctive ring has been fixed to work for queue members -- rtp -- Fixed a logic error when setting the "rtpchecksums" option -- say.c -- A problem has been fixed with saying the date in Spanish. -- Makefile -- A line was missing for the autosupport script that caused "make rpm" to fail -- format_wav_gsm -- Fixed a problem with wav formatting that prevented files from being played in some media players -- pbx_spool -- Fixed if the last line of text in a file for the call spool did not contain a new line, it would not be processed -- logger -- Fixed the logger so that color escape sequences wouldn't be sent to the logs -- format_sln -- A lot of changes were made to correctly handle signed linear format on big endian machines
2005-09-02 14:58:34 +02:00
@dirrm share/examples/asterisk
Update asterisk to version 1.2.1. Many, many bugfixes, and some new features, including support for DUNDi. (http://www.dundi.com/ for more information) The initial framework and porting of this package upgrade was done by Martin J. Laubach, with lots of feature/PLIST fixes by me. DragonFly support added by Joerg Sonnenberger.
2006-01-13 21:32:38 +01:00
@dirrm libdata/asterisk/sounds/priv-callerintros
Changes 1.0.9: -- fix bug in callerid matching in the dialplan that was introduced in 1.0.8 Changes 1.0.8: -- chan_zap -- Asterisk will now also look in the regular context for the fax extension while executing a macro. Previously, for this to work, the fax extension would have to be included in the macro definition. -- On some systems, ALERTING will be sent after PROCEEDING, so code has been added to account for this case. -- If no extension is specified on an overlap call, the 's' extension will be used. -- chan_sip -- We no longer send a "to" tag on "100 Trying" messages, as it is inappropriate to do so. -- We now respond correctly to an invite for T.38 with a "488 Not acceptable here" -- We now discard saved tags on 401/407 responses in case the provider we're talking to tries to pull a dirty trick on us and change it. -- rtptimeout options will now be correctly set on a peer basis rather than only global -- chan_mgcp -- Fixed setting of accountcode -- Fixed where *67 to block callerid only worked for first call -- chan_agent -- We now will not pass audio until the agent has acked the call if the configuration is set up for the agent to do so. -- chan_alsa -- Fixed problems with the unloading of this module -- res_agi -- A fix has been added to prevent calls from being hung up when more than one call is executing an AGI script calling the GET DATA command. -- AGI scripts will now continue to run even if a file was not found with the GET DATA command. -- When calling SAY NUMBER with a number like 09, we will now say "nine" instead of "zero" -- app_dial -- There was a problem where text frames would not be forwarded before the channel has been answered. -- app_disa -- Fixed the timeout used when no password is set -- app_queue -- Distinctive ring has been fixed to work for queue members -- rtp -- Fixed a logic error when setting the "rtpchecksums" option -- say.c -- A problem has been fixed with saying the date in Spanish. -- Makefile -- A line was missing for the autosupport script that caused "make rpm" to fail -- format_wav_gsm -- Fixed a problem with wav formatting that prevented files from being played in some media players -- pbx_spool -- Fixed if the last line of text in a file for the call spool did not contain a new line, it would not be processed -- logger -- Fixed the logger so that color escape sequences wouldn't be sent to the logs -- format_sln -- A lot of changes were made to correctly handle signed linear format on big endian machines
2005-09-02 14:58:34 +02:00
@dirrm libdata/asterisk/sounds/phonetic
@dirrm libdata/asterisk/sounds/letters
@dirrm libdata/asterisk/sounds/digits
Update asterisk to version 1.2.1. Many, many bugfixes, and some new features, including support for DUNDi. (http://www.dundi.com/ for more information) The initial framework and porting of this package upgrade was done by Martin J. Laubach, with lots of feature/PLIST fixes by me. DragonFly support added by Joerg Sonnenberger.
2006-01-13 21:32:38 +01:00
@dirrm libdata/asterisk/sounds/dictate
Update asterisk to 1.2.14.
2006-12-20 12:34:55 +01:00
@dirrm libdata/asterisk/sounds/silence
Changes 1.0.9: -- fix bug in callerid matching in the dialplan that was introduced in 1.0.8 Changes 1.0.8: -- chan_zap -- Asterisk will now also look in the regular context for the fax extension while executing a macro. Previously, for this to work, the fax extension would have to be included in the macro definition. -- On some systems, ALERTING will be sent after PROCEEDING, so code has been added to account for this case. -- If no extension is specified on an overlap call, the 's' extension will be used. -- chan_sip -- We no longer send a "to" tag on "100 Trying" messages, as it is inappropriate to do so. -- We now respond correctly to an invite for T.38 with a "488 Not acceptable here" -- We now discard saved tags on 401/407 responses in case the provider we're talking to tries to pull a dirty trick on us and change it. -- rtptimeout options will now be correctly set on a peer basis rather than only global -- chan_mgcp -- Fixed setting of accountcode -- Fixed where *67 to block callerid only worked for first call -- chan_agent -- We now will not pass audio until the agent has acked the call if the configuration is set up for the agent to do so. -- chan_alsa -- Fixed problems with the unloading of this module -- res_agi -- A fix has been added to prevent calls from being hung up when more than one call is executing an AGI script calling the GET DATA command. -- AGI scripts will now continue to run even if a file was not found with the GET DATA command. -- When calling SAY NUMBER with a number like 09, we will now say "nine" instead of "zero" -- app_dial -- There was a problem where text frames would not be forwarded before the channel has been answered. -- app_disa -- Fixed the timeout used when no password is set -- app_queue -- Distinctive ring has been fixed to work for queue members -- rtp -- Fixed a logic error when setting the "rtpchecksums" option -- say.c -- A problem has been fixed with saying the date in Spanish. -- Makefile -- A line was missing for the autosupport script that caused "make rpm" to fail -- format_wav_gsm -- Fixed a problem with wav formatting that prevented files from being played in some media players -- pbx_spool -- Fixed if the last line of text in a file for the call spool did not contain a new line, it would not be processed -- logger -- Fixed the logger so that color escape sequences wouldn't be sent to the logs -- format_sln -- A lot of changes were made to correctly handle signed linear format on big endian machines
2005-09-02 14:58:34 +02:00
@dirrm libdata/asterisk/sounds
@dirrm libdata/asterisk/mohmp3
@dirrm libdata/asterisk/keys
@dirrm libdata/asterisk/images
@dirrm libdata/asterisk/firmware/iax
@dirrm libdata/asterisk/firmware
@dirrm libdata/asterisk/agi-bin
@dirrm libdata/asterisk
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
@dirrm lib/asterisk/modules
@dirrm lib/asterisk
@dirrm include/asterisk
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