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Changes since Asterisk 1.2: |
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* over 4,000 commits since 1.2 |
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* queue member naming |
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* CLI commands rework |
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o Change the way CLI commands are structured. |
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o Most commands are now <module> <verb> <args> |
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* chan_h323 update |
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* RTP packetization |
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* SLA (Shared Line Appearance) support |
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* T.38 Passthrough Support for faxing in SIP |
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* Generic channel jitterbuffer (spawned from RTP) |
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* Variable Length DTMF for better DTMF compatibility |
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* Improved chan_iax2 scalability by using multithreading |
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* AEL2 has replaced the original implementation of AEL. The "2" is removed. For more details, |
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read: http://www.voip-info.org/wiki/view/Asterisk+AEL2 |
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AEL is no longer considered experimental. |
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* New sounds; English, Spanish, and French prompts, as well as music on hold files, in |
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multiple Asterisk native formats. |
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* IMAP storage of voicemail |
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* Jabber/GoogleTalk integration |
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* New speech recognition API for interfacing to different Voice Recognition software packages |
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* much more customizable and portable build system |
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o also for asterisk-addons |
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* Radius CDR logging |
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* SNMP support |
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* SMDI (Simplified Message Desk Interface) support |
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* Redesign of MusicOnHold configuration settings |
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* Manager over HTTP |
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* Significant chan_skinny updates |
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* Significant chan_misdn updates |
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* Improved SIP transfers |
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* SIP MWI subscription support |
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* Much improved support for SIP video |
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* Control over SIP transfers and subscriptions (enable/disable per device) |
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* ChanSpy whisper mode (Whisper Paging) |
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* Configurable language support for saying dates and times |
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* Significant architecture improvements for memory usage and performance |
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* Media-only IAX2 transfers |
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* Updates to the Radio Repeater app code |
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* Deprecation of AgentCallbackLogin in favor of a dialplan-based solution |
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* uClibc builds supported |
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* Work done for freeBSD portability |
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* Work done for Solaris portability |
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* FreeTDS-based database can be used with Realtime |
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* New internal data structure, stringfields, is implemented in IAX and SIP, reducing memory consumption by about 50%. |
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* Use of thread local storage for reduced memory allocation/freeing and lower stack consumption |
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* Reorganized files into docs/ main/ configs/, including name changes in some cases |
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* Much effort was expended in arranging documentation in source files in doxygen format |
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* Improved IP TOS support for IAX and SIP |
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* Builtin mini HTTP server |
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* Added support for Sigma Designs cards. |
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* Frame header caching to reduce memory allocation/freeing |
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* Passthrough and record/playback support for G.722 wideband audio |
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* using mpg123 to play MP3 files for music-on-hold will be deprecated in 1.4 (start using the "native support") |
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* New Apps: |
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1. AMD() ;; Answering Machine Detection |
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2. ChannelRedirect() ;; asynch goto, redirect chan to context/exten/priority |
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3. ContinueWhile() ;; Addition to the While() suite. Acts like "continue". |
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4. ExitWhile() ;; Addition to the While() suite. Acts like "break". |
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5. ExtenSpy() ;; A close cousin to ChanSpy(). |
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6. FollowMe() ;; findme/followme call redirect app |
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7. Log() ;; Send a message to the log, based on severity level. |
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8. MacroExclusive() ;; No more than one invocation of this macro allowed at any one time. |
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9. MorseCode() ;; turns strings into dits and dahs. A playground for ham radio licensees! |
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10. OSPAuth() ;; OSP authentication |
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11. QueueLog() ;; allows you to write your own events into the queue log |
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12. SLAStation() ;; Shared Line Appearance |
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13. SLATrunk() ;; Shared Line Appearance |
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14. SpeechCreate() ;; Voice Recognition Engine interface... |
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15. SpeechActivateGrammar() |
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16. SpeechStart() |
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17. SpeechBackground |
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18. SpeechDeactivateGrammar() |
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19. SpeechProcessingSound() |
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20. SpeechDestroy() |
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21. SpeechLoadGrammar() |
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22. SpeechUnloadGrammar() |
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23. StopMixMonitor() ;; to stop the MixMonitor App. |
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24. TryExec() ;; execute dialplan app without fatal consequences |
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* Apps removed: |
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1. CheckGroup -- do a comparison to ${GROUP()} |
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2. Curl -- use the function CURL() instead |
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3. Cut -- use the function CUT() instead |
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4. DateTime -- use sayunixtime() app instead. |
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5. DBget -- deprecated in 1.2, now removed. |
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6. DBput -- deprecated in 1.2, now removed. |
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7. Enumlookup -- use the function ENUMLOOKUP() instead |
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8. Eval -- use the function EVAL() instead |
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9. GetGroupCount -- use the function GROUP_COUNT() instead |
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10. GetGroupMatchCount -- use the function GROUP_MATCH_COUNT() instead |
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11. Intercom -- use the chan_oss module instead |
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12. Math -- use the function MATH() instead |
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13. MD5 -- use the function MD5() instead |
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14. SetCIDname -- use the function CALLERID(name) instead |
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15. SetCIDnum -- use the function CALLERID(number) instead |
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16. SetGroup -- use Set(GROUP=group) instead |
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17. SetRDNIS -- use the function CALLERID(rdnis) instead |
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18. Sql_postgres -- was deprecated in 1.2, now removed |
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19. Txtcidname -- use the function TXTCIDNAME instead |
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* New Dialplan Functions: |
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1. ARRAY() |
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2. BASE_64_DECODE() |
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3. BASE_64_ENCODE() |
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4. CHANNEL() |
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5. CURL() |
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6. CUT() |
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7. DB_DELETE() |
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8. FILTER() |
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9. GLOBAL() |
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10. IFTIME() |
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11. KEYPADHASH() |
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12. ODBC() |
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13. QUOTE() |
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14. RAND() |
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15. REALTIME() |
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16. SHA1() |
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17. SORT() |
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18. SPRINTF() |
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19. SQL_ESC() |
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20. STAT() |
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21. STRPTIME() |
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* Apps that have changes to their interface: |
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1. Authenticate() -- optional maxdigits argument added. |
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2. ChanSpy() -- new options: |
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o w -- Enable 'whisper' mode, so the spying channel can talk to... |
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o W -- Enable 'private whisper' mode, so the spying channel can... |
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3. DBdel() -- now marked as DEPRECATED in favor of the DB_DELETE func |
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4. Dial() |
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o New Option: O([x]) for Zaptel operator mode |
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o New Option: K/k parking via dtmf tones |
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5. Dictate() -- optional filename argument added. |
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6. Directory() -- new option: e - In addition to the name, also read the extension number... |
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7. Meetme() -- new options: |
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o 'I' -- announce user join/leave without review |
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o 'l' -- set listen only mode (Listen only, no talking) |
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o 'o' -- set talker optimization - treats talkers who aren't speaking as... |
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o '1' -- do not play message when first person enters |
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8. MeetmeAdmin() -- new options: |
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o 'r' -- Reset one user's volume settings |
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o 'R' -- Reset all users volume settings |
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o 's' -- Lower entire conference speaking volume |
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o 'S' -- Raise entire conference speaking volume |
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o 't' -- Lower one user's talk volume |
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o 'T' -- Lower all users talk volume |
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o 'u' -- Lower one user's listen volume |
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o 'U' -- Lower all users listen volume |
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o 'v' -- Lower entire conference listening volume |
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o 'V' -- Raise entire conference listening volume |
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9. OSPFinish() : now also can return ERROR result. |
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10. OSPLookup() : Sets more variables, also now returns ERROR result. |
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11. Page() -- New option: r - record the page into a file (see 'r' for app_meetme) |
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12. Pickup() -- multiple extensions, PICKUPMARK; read the description! |
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13. Queue() |
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o New Argument: AGI |
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o New option: i |
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14. Random() -- is now deprecated in 1.4 |
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15. Read() -- replace 'skip' and 'noanswer' options with 's', 'n', add 'i' option. |
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16. Record() -- New option: 'x' : ignore all terminator keys (DTMF) and keep recording until hangup |
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17. UserEvent() -- slight change in behavior. Read the description. |
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18. VoiceMailMain() -- new a(#) option, goes to folder # directly. |
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19. WaitForSilence() -- new optional 3rd arg, time delay before returning. |
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* Functions that have changes to their interfaces: |
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1. CDR -- new option: u |
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2. LANGUAGE -- Deprecated. Use CHANNEL(language) instead. |
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3. MUSICCLASS -- Deprecated. Use CHANNEL(musicclass) instead. |
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* Configuration File Changes: |
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1. NEW config files: |
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1. amd.conf -- Answering Machine Detection parameters |
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2. followme.conf -- parameters for the findme/followme call forwarding |
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3. func_odbc.conf -- define sql access functions here |
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4. gtalk.conf -- how to handle gtalk protocol calls |
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5. h323.conf -- h323 configuration |
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6. http.conf -- config for the builtin mini-http server in asterisk |
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7. jabber.conf -- jabber interface |
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8. jingle.conf -- jingle protocol interface config |
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10. res_snmp.conf -- to enable snmp in asterisk, and define full/sub agent status |
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11. say.conf -- define per-language rules for numbers, dates, etc. |
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12. skinny.conf -- for those special skinny phones you want to use... |
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13. sla.conf -- Shared Line Appearance config |
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14. smdi.conf -- SMDI messaging config |
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15. udptl.conf -- T38's udptl transport config |
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16. users.conf -- user config |
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2. Changes to Existing Config files: |
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1. In General: |
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o Jitterbuffer support added to several channels. Usually adds these variables to a config file: |
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1. jbenable |
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2. jbmaxsize |
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3. jbresyncthreshold |
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4. jbimpl |
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5. jblog |
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o MusicOnHold upgrade introduces two new variables: |
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1. mohinterpret |
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2. mohsuggest |
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2. agents.conf |
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o maxlogintries variable added |
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o autologoffunavail variable added |
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o endcall variable added |
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o agentgoodbye variable added |
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o createlink variable REMOVED |
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3. alsa.conf |
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o mohinterpret variable added |
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o Jitterbuffer variables added |
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4. cdr.conf |
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o endbeforehexten variable added |
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o sections for csv and radius added, with variables usegmtime, loguniqueid, |
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loguserfield, and radiuscfg variables. |
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5. cdr_tds.conf |
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o table variable added |
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6. extensions.ael |
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o Many upgrades. See the info at http://www.voip-info.org/wiki/view/Asterisk+AEL2 |
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7. extensions.conf |
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o autofallthru now set to "yes" by default |
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o userscontext variable added |
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o added info/examples on paging and hints. |
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8. features.conf |
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o parkedplay variable added (who to beep at) |
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o parkedmusicclass |
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o atxfernoanswertimeout variable added |
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o parkcall variable added (one step parking) |
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o improved documentation for dynamic feature declarations! |
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9. iax.conf |
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o adsi variable added |
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o mohinterpret variable added |
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o mohsuggest variable added |
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o jitterbuffer updates |
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o iaxthreadcount variable added |
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o iaxmaxthreadcount variable added |
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o the way to specify TOS has changed. |
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o mailboxdetail variable has been REMOVED. |
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10. indications.conf |
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o [bg] entry added (Bulgaria). |
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o [il] entry added (Israel) |
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o [in] entry added (India) |
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o [jp] entry added (Japan) |
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o [my] entry added (Malaysia) |
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o [th] entry added (Thailand) |
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11. manager.conf |
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o webenabled variable added |
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o httptimeout variable added |
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o timestampevents variable added |
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12. mgcp.conf |
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o Jitterbuffer support added |
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13. misdn.conf |
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o l1watcher_timeout variable added |
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o pp_l2_check variable added |
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o echocancelwhenbridged variable added |
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o echotraining variable added |
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o max_incoming variable added |
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o max_outgoing variable added |
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14. modules.conf |
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o a comment for preloading res_speech.so is added |
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o mention of global symbols is removed |
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o obsolesced entries for chan_modem_* and app_intercom have been removed |
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15. musiconhold.conf |
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o the default is now to do native moh from /var/lib/asterisk/moh |
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16. osp.conf |
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o authpolicy variable added |
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17. oss.conf |
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o debug variable added |
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o device variable added |
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o mixer variable added |
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o boost variable added |
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o callerid variable added |
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o autohangup variable added |
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o queuesize variable added |
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o frags variable added |
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o JitterBuffer support |
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o sections to define alternate sound cards |
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18. queues.conf |
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o autofill variable added |
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o monitor-type variable added |
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o musiconhold is now musicclass, with a difference in interpretation |
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o autofill variable added |
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o autopause variable added |
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o setinterfacevar variable added |
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o ringinuse variable added |
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19. res_odbc.conf |
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o pooling variable added |
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20. rpt.conf |
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o duplex variable added |
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o tailmessagetime variable added |
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o tailsquashedtime variable added |
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o tailmessages variable added |
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21. rtp.conf |
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o rtcpinterval varaible added |
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22. sip.conf |
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o allowoverlap variable added |
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o allowtransfer variable added |
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o tos variable REMOVED |
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o tos_sip variable added |
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o tos_audio variable added |
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o tos_video variable added |
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o minexpiry variable added |
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o t1min variable added |
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o musicclass variable REMOVED |
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o mohinterpret variable added |
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o maxcallbitratesuggest variable added |
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o allowsubscribe variable added |
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o videosupport variable added |
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o maxcallbitrate variable added |
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o g726nonstandard variable added |
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o dumphistory variable added |
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o allowsubscribe variable added |
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o t38pt_udptl variable added |
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o canreinvite variable can also now be set to 'nonat' |
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o rtsavesysname variable added |
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o JitterBuffer support added |
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23. skinny.conf |
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o port variable renamed to bindport |
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o JitterBuffer support added |
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o model variable REMOVED |
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o mohinterpret variable added |
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o mohsuggest variable added |
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o speeddial variable added |
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o addon variable added |
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24. voicemail.conf |
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o userscontext variable added |
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o smdiport variable added |
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o attachfmt variable added |
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o volgain variable added |
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o tempgreetwarn variable added |
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25. zapata.conf |
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o pritimer variable has improved documentation |
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o New signalling method: fgccama |
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o New signalling method: fgccamamf |
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o outsignalling variable added |
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o distinctiveringaftercid variable added |
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o cidsignalling now also accepts v23_jp, and smdi |
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o usesmdi variable added |
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o smdiport variable added |
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o mohinterpret variable added |
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o mohsuggest variable added |
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o JitterBuffer support added |
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* Removed Codecs/Channels: |
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1. codec_g723 was removed because the actual codec implementation it was designed to use is not distributable |
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2. chan_modem_* and related modules are gone because the kernel support for those interfaces is old, buggy and unsupported |
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* New Utils: |
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1. aelparse -- compile .ael files outside of asterisk |
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* New manager events: |
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1. OriginateResponse event comes to replace OriginateSuccess and OriginateFailure |
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