pkgsrc/comms/asterisk18/distinfo

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$NetBSD: distinfo,v 1.3 2010/12/22 04:28:52 jnemeth Exp $
Import Asterisk 1.8.1: 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 1.8 is a long term support version (i.e. it will be supported for four years with an additional year of security only fixes). See: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions What's new: Asterisk 1.8 is the next major release series of Asterisk. The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we've had over 500 reporters, more than 300 testers and greater than 200 developers contributed to this release. You can find a summary of the work involved with the 1.8.0 release in the sumary: http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt A short list of available features includes: * Secure RTP * IPv6 Support in the SIP channel driver * Connected Party Identification Support * Calendaring Integration * A new call logging system, Channel Event Logging (CEL) * Distributed Device State using Jabber/XMPP PubSub * Call Completion Supplementary Services support * Advice of Charge support * Much, much more! A full list of new features can be found in the CHANGES file. http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup For a full list of changes in the current release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0 ----- The Asterisk Development Team has announced the release of Asterisk 1.8.1. The release of Asterisk 1.8.1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix issue when using directmedia. Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (Closes issue #17403. Reported, patched by one47. Tested by one47, falves11) * Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers. (Patched by rmudgett) * Fix playback failure when using IAX with the timerfd module. (Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) * Fix issue where it is possible to crash Asterisk by feeding the curl engine invalid data. (Closes issue #18161. Reported by wdoekes. Patched by tilghman) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
2010-12-15 04:22:43 +01:00
SHA1 (asterisk-1.8.1.1/asterisk-1.8.1.1.tar.gz) = 8499778d83a8f35ce5abbfe4680de1701dc10f7c
RMD160 (asterisk-1.8.1.1/asterisk-1.8.1.1.tar.gz) = a761369083f9b5789dd055ab7df678b1cce83287
Size (asterisk-1.8.1.1/asterisk-1.8.1.1.tar.gz) = 26198619 bytes
SHA1 (asterisk-1.8.1.1/asterisk-extra-sounds-en-gsm-1.4.11.tar.gz) = 8692fa61423b4769dc8bfa78faf9ed5ef7a259b9
RMD160 (asterisk-1.8.1.1/asterisk-extra-sounds-en-gsm-1.4.11.tar.gz) = 68170c769d739d6b5b35b00f999ad6bbf876f9f6
Size (asterisk-1.8.1.1/asterisk-extra-sounds-en-gsm-1.4.11.tar.gz) = 3349898 bytes
SHA1 (asterisk-1.8.1.1/extract-cfile.awk) = c4f08eee1ab83c041bde1ab91672a4a3c43c28b8
RMD160 (asterisk-1.8.1.1/extract-cfile.awk) = cd59f8e5807732023d5aec95187e2d5572f400a4
Size (asterisk-1.8.1.1/extract-cfile.awk) = 667 bytes
SHA1 (asterisk-1.8.1.1/rfc3951.txt) = 1a6c769be750fb02456d60db2470909254496017
RMD160 (asterisk-1.8.1.1/rfc3951.txt) = 15f7ec61653ec9953172f8f2150e7d8f6f620926
Size (asterisk-1.8.1.1/rfc3951.txt) = 373442 bytes
Import Asterisk 1.8.1: 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 1.8 is a long term support version (i.e. it will be supported for four years with an additional year of security only fixes). See: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions What's new: Asterisk 1.8 is the next major release series of Asterisk. The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we've had over 500 reporters, more than 300 testers and greater than 200 developers contributed to this release. You can find a summary of the work involved with the 1.8.0 release in the sumary: http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt A short list of available features includes: * Secure RTP * IPv6 Support in the SIP channel driver * Connected Party Identification Support * Calendaring Integration * A new call logging system, Channel Event Logging (CEL) * Distributed Device State using Jabber/XMPP PubSub * Call Completion Supplementary Services support * Advice of Charge support * Much, much more! A full list of new features can be found in the CHANGES file. http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup For a full list of changes in the current release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0 ----- The Asterisk Development Team has announced the release of Asterisk 1.8.1. The release of Asterisk 1.8.1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix issue when using directmedia. Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (Closes issue #17403. Reported, patched by one47. Tested by one47, falves11) * Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers. (Patched by rmudgett) * Fix playback failure when using IAX with the timerfd module. (Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) * Fix issue where it is possible to crash Asterisk by feeding the curl engine invalid data. (Closes issue #18161. Reported by wdoekes. Patched by tilghman) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
2010-12-15 04:22:43 +01:00
SHA1 (patch-aa) = a157fe745bde7880cbbdcfdf9e4bb4381f1df185
SHA1 (patch-af) = ebad62fcb31b600d30235cc5e93284c93b2c8af9
SHA1 (patch-ag) = c71c61350cefbbe53eefa99245ca7712753f22d5
SHA1 (patch-ai) = e92edab5c1ff323478f41d0b0783102ed527fe39
SHA1 (patch-ak) = adee75b7716a8794de1b8cb054af7a5a8f0e5ffd
SHA1 (patch-al) = b2a1134786d7c3b118ee8c47892f91dd2a4c783a
SHA1 (patch-am) = 5f9cbf47ec1cb66758492a5ed1bf843006eae9b7
SHA1 (patch-an) = 93a5df66fd6459fb76e9191dc3bf37b9ee5483b5
SHA1 (patch-ao) = 0663a698469550b22bb97ee1b18980bc2bc67495
SHA1 (patch-ap) = c36531be80784d47c2c772ccc386ef2b8f71c72f
SHA1 (patch-aq) = 9e05e8d099b92f6c087ca083c5a6d89a0c345061
SHA1 (patch-ar) = 5e117f173e6743703d5d83787877e7a9ce8500f3
SHA1 (patch-as) = 84b84acac731671944981a60124d0447a2cd322c
Import Asterisk 1.8.1: 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 1.8 is a long term support version (i.e. it will be supported for four years with an additional year of security only fixes). See: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions What's new: Asterisk 1.8 is the next major release series of Asterisk. The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we've had over 500 reporters, more than 300 testers and greater than 200 developers contributed to this release. You can find a summary of the work involved with the 1.8.0 release in the sumary: http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt A short list of available features includes: * Secure RTP * IPv6 Support in the SIP channel driver * Connected Party Identification Support * Calendaring Integration * A new call logging system, Channel Event Logging (CEL) * Distributed Device State using Jabber/XMPP PubSub * Call Completion Supplementary Services support * Advice of Charge support * Much, much more! A full list of new features can be found in the CHANGES file. http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup For a full list of changes in the current release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0 ----- The Asterisk Development Team has announced the release of Asterisk 1.8.1. The release of Asterisk 1.8.1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix issue when using directmedia. Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (Closes issue #17403. Reported, patched by one47. Tested by one47, falves11) * Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers. (Patched by rmudgett) * Fix playback failure when using IAX with the timerfd module. (Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) * Fix issue where it is possible to crash Asterisk by feeding the curl engine invalid data. (Closes issue #18161. Reported by wdoekes. Patched by tilghman) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
2010-12-15 04:22:43 +01:00
SHA1 (patch-at) = ce094dc436cc4fd6aeafca3460a25c2db077eaf8
SHA1 (patch-au) = 57100ee55338c5ab8be1f8d73d4126d26227786b
SHA1 (patch-av) = 3424013b5bf22624aa664e972e2b495ab3296cbe
SHA1 (patch-aw) = 0534acd67ea5da1eee8cf282035ebf4c559278ab
SHA1 (patch-ax) = 3b41e66a8c926e0afc4f73587e3557370e6c5f6e
SHA1 (patch-ay) = 824fc560f4f2775ecf9272525025d26d8fee4361
SHA1 (patch-az) = 64365b12cb47ec0fba358e4326eda172f96068cf
SHA1 (patch-ba) = ffb20f4788f2f253e822fb48c68fec04c31b0619
SHA1 (patch-bb) = bf1a2bb2ba1eb2ba44a9b26fa9ae0468510a1575
Import Asterisk 1.8.1: 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 1.8 is a long term support version (i.e. it will be supported for four years with an additional year of security only fixes). See: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions What's new: Asterisk 1.8 is the next major release series of Asterisk. The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we've had over 500 reporters, more than 300 testers and greater than 200 developers contributed to this release. You can find a summary of the work involved with the 1.8.0 release in the sumary: http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt A short list of available features includes: * Secure RTP * IPv6 Support in the SIP channel driver * Connected Party Identification Support * Calendaring Integration * A new call logging system, Channel Event Logging (CEL) * Distributed Device State using Jabber/XMPP PubSub * Call Completion Supplementary Services support * Advice of Charge support * Much, much more! A full list of new features can be found in the CHANGES file. http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup For a full list of changes in the current release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0 ----- The Asterisk Development Team has announced the release of Asterisk 1.8.1. The release of Asterisk 1.8.1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix issue when using directmedia. Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (Closes issue #17403. Reported, patched by one47. Tested by one47, falves11) * Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers. (Patched by rmudgett) * Fix playback failure when using IAX with the timerfd module. (Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) * Fix issue where it is possible to crash Asterisk by feeding the curl engine invalid data. (Closes issue #18161. Reported by wdoekes. Patched by tilghman) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
2010-12-15 04:22:43 +01:00
SHA1 (patch-bc) = 0efc99595d1ef82a879361e8bf3b2ef7fd84af62
SHA1 (patch-be) = a3d416c097c6aeb0e49dec67a9fc22027d936773
SHA1 (patch-bf) = 67b506d235fabaa73f492d08858407dd9a85fd6e
SHA1 (patch-bg) = 07df551999f8e33db7bb613f666626de8be3036c
SHA1 (patch-bh) = 9203ea97daab8c64ea47f236b4961763e76eafe6
SHA1 (patch-bi) = d71662f618a10c3ca4277feb7ad0d659935dee1e
SHA1 (patch-bj) = a184452adf2c883695e3819c13c584a3db9608d7
SHA1 (patch-bk) = 93679dfb04d26c99ac9c2822e0d74d869d16369f