2.5.6 -> 2.5.7 (hbfl) 2019-03-17 ================================ - Anlegen von Gruppe und User ueberarbeitet 2.5.5 -> 2.5.6 (hbfl) 2017-07-11 ================================ - Check fuer ASTERISK_AREA_CODE korrigiert (Stefan Welte) 2.5.4 -> 2.5.5 (hbfl) 2017-04-15 ================================ - Tippfehler in chmod fuer das Device capi20 behoben. Kernel-3.16 2.5.3 -> 2.5.4 (hbfl) 2017-03–23 ================================ - Anpassung an CAPI mit udev (Benjamin Heide) 2.5.2 -> 2.5.3 (hbfl) 2016-09-07 ================================ - Anpassungen an den neuen Installer 'eisman' 2.5.1 -> 2.5.2 (hbfl) 2016-04-15 ================================ - dahdi Kontrollskript wird beim Start von Asterisk entfernt. Zirkualerer start. (Stefan Welte) 2.5.0 -> 2.5.1 (hbfl) 2016-04-14 ================================ - einige Typos behoben. 2.1.13 -> 2.5.0 (hbfl) 2015-12-29 ================================= - Unterstuetzung fuer SCCP Telefone hinzugefuegt Dank an Stephan Sanders fuer den diff. - update Asterisk Version 1.8.32.3 - update asterisk_httpd Version 1.23 2.1.12 -> 2.1.13 (hbfl) 2015-04-06 ================================== - Neuer Parameter: ASTERISK_WEBINTERFACE_FORCE_SSL Asterisk startet nur noch eine Webinterface Instanz die durch ASTERISK_WEBINTERFACE_FORCE_SSL behandelt wird. Standard ist 'yes' 2.1.11 -> 2.1.12 (hbfl) 2015-03-16 ================================== - Fehler durch deinstall beim Update behoben (Stefan Welte). /var/run/${package_name}.ctl /var/log/${package_name}/ /var/spool/${package_name}/ Die Verzeichnisse werden jetzt nur bei einer entgueltigen Deinstallation entfernt. - Das Webinterface laeuft nun mit charset UTF-8 - Fehler zum korrekten starten des Webinterface behoben 'force-cgi-redirect' (Stefan Welte). - PHP Fehler im Webinterface behoben - Asterisk benutzt nun den asterisk_httpd - Wird bei einem update eine 'httpd.pem' Datei gefunden so wird diese automatisch in die Konfiguration eingebunden. Damit startet das Webinterface mit SSL sofort wieder. 2.1.10 -> 2.1.11 (hbfl) 2015-03-11 ================================== - Weitere Asterisk Varibalen ueber Paramter konfigurierbar gemacht (Stefan Welte). ASTERISK_SIP_PORT ASTERISK_SIP_QUALIFY ASTERISK_IAX_QUALIFY ASTERISK_LOCALNETS_N - Asterisk verwendet nun das Standard SSL-Server-Zertifikat ASTERISK_WEBINTERFACE_CERTIFICATE Beim update wird das Webinterface nicht automatisch wieder gestartet. - Asterisk configs koennen nun auch per Hand erstellt werden (Dirk Alberti). ASTERISK_CONFIG_USE -> default 'yes' 2.1.9 -> 2.1.10 (hbfl) 2015-02-17 ================================= - Fehler im parsen der config mit PHP behoben, Dank an Daniel Vogel. - update Asterisk Version 1.8.32.2 - das Asterisk Webinterface wurde nach /var/lib/asterisk/webinterface verschoben 2.1.8 -> 2.1.9 (hbfl) 2014-12-17 ================================ - weitere Syntax anpassungen in extensions.conf '|' in ',' 2.1.7 -> 2.1.8 (hbfl) 2014-12-09 ================================ - Fehler im Debug Menue behoben - Pruefen ob die Devices /dev/dahdi/* Asterisk gehoeren - Geaenderte Syntax fuer {Digit,Response}Timeout in CALLTHROUGH 2.1.6 -> 2.1.7 (hbfl) 2014-12-08 ================================ - Beim Vorfinden einer alten Asterisk Version <= 2.1.7 wird die config gesichert, Asterisk deinstalliert, die config zurueck geschrieben und Asterisk installiert. Um Probleme mit neuen und alten Modulen zu beheben. 2.1.5 -> 2.1.6 (hbfl) 2014-12-07 ================================ - PHP-fehler in CALLTHROUGH und VBOX behoben 2.1.4 -> 2.1.5 (hbfl) 2014-11-30 ================================ - start/stop Skripte fuer capi und dahdi hinzugefuegt - config-SIP defaultexpirey=300 2.1.3 -> 2.1.4 (hbfl) 2014-11-29 ================================ - Scheck fuer chan-dahdi.so hinzugfuegt - Anpassungen an Dahdi - Asterisk Syntax anpassungen 2.1.2 -> 2.1.3 (hbfl) 2014-11-25 ================================ - Scheck fuer chan_capi.so hinzugefuegt. - Das Verzeichnis /usr/lib/asterisk/modules wird beim update nicht mehr entfernt. 2.1.1 -> 2.1.2 (hbfl) 2014-11-24 ================================ - diverse Korrekturen in den PHP Skripten - Asterisk Syntax anpassungen 2.1.0 -> 2.1.1 (hbfl) 2014-11-20 ================================ - Dateiberechtigungen fuer asterisk gesetzt - diverse deprecated Warnungen ersetzt 1.9.1 -> 2.1.0 (hbfl) 2014-10-23 ================================ - status unstable - update Asterisk Version 1.8.30 - Fuer Telephonekarten wird Dahdi benoetigt und erforderlich ist eiskernel-2.5.x 1.9.9 -> 1.9.1 (hbfl) 2013-07-26 ================================= - Behandlung von HFC-S Karten im TE Modus korrigiert (Piotr Kujawski) 1.3.4 -> 1.9.0 (hbfl) 2013-03-22 ================================ - update bristuff-0.3.0-PRE-1z-e - Extra Module: chan_sccp chan_unistim chan_capi - Module fuer Kernel 2.6.32-eisfair-1-SMP Module fuer Kernel 2.6.32-eisfair-1-PAE 1.3.3 -> 1.3.4 (hbfl) 2012-08-26 ================================ - status testing 1.3.2 -> 1.3.3 (hbfl) 2012-05-14 ================================ - Aenderungen in den PHP-Skripten rueckgaengig gemacht - error-reporting bleibt gesetzt 1.3.1 -> 1.3.2 (hbfl) 2012-04-20 ================================ - PHP error-reporting(E_ERROR & E_WARNING) in der Applikation gesetzt, um einen Start zu ermoeglichen, bis alle PHP Meckereien eingefangen wurden. 1.3.0 -> 1.3.1 (hbfl) 2012-04-09 ================================ - schreiben der Konfigdatei erfolgt nun mit asterisk-update.sh und configlib - Fehler im include codec_speex behoben(Roland Wieland) 1.1.9 -> 1.3.0 (hbfl) 2012-04-04 ================================ - Status unstable - update bristuff-0.3.0-PRE-1z-d Asterisk 1.2.40 Module fuer Kernel 2.6.32-eisfair-1{-SMP} asterisk 1.1.8 -> 1.1.9 2006/10/24 testing =========================================== - fixed broken MySQL connect in reverse lookup script - fixed webinterface to suit php5_ccpp instead of php5_cgi - removed pseudo configuration from webinterface - fixed bug in webinterface protocol module. Wrong allocation between overview and detailed view - reloading all modules at startup, also if zaphfc is not loaded asterisk 1.1.7 -> 1.1.8 2006/09/29 testing =========================================== - updated bristuff to 0.3.0-PRE-1r - IVR in conjunction with redirect schemes works - fixed dasoertliche.de (new layout) asterisk 1.1.6 -> 1.1.7 2006/07/28 testing =========================================== - MySQL Database for phonebook data is created automatically if it doesn't exist - fixed bug that causes recorded conversations not to play back in the webinterface - changed channel unavailable announcement to congestion tone - webinterface: moved call playback from call overview to detail site - removed tt-somethingwrong in lcr script when falling back - fixed dasoertliche.de reverselookup - changed status to testing asterisk 1.1.5 -> 1.1.6 2006/07/05 unstable =========================================== - fixed bug when using LCR and a CAPI dialprefix. The default area code was prefixed in front of the dialed area code - capi dial outs without using LCR weren't possible asterisk 1.1.4 -> 1.1.5 2006/07/04 unstable =========================================== - added ACTIVE variables in array configuration paragraphes - fixed warning when stopping asterisk and fixed bug that vmail webinterface didn't work - more than one webinterface user is possible - replaced busy and congestion announcements by tones - fixed echotest - fixed error message in voicemail main menu: Unable to create lock file - fixed bug in voice mail main menu that "oclock" ("Uhr") is played twice - more than one email address in vbox settings possible - callback: trying two times to callback, instead of one time - callback: if running in a timeout, the entered number will be dialed (same with PIN) - callback: using LCR - SIP: dtmfmode is now auto - improved handling of unkown numbers in reverse lookup script and other things - reverse lookup is now possible in conjunction with a mysql database - remote reverse lookup can be disabled - MWI is working for SIP - incoming calls can be signalised over Netbios - added option that playing back the vbox is possible with dialing only the vbox dialprefix - disabled NAT in sip.conf for peers. Enabled NAT in global context. - added defaultexpirey in sip.conf for t-online - telephone conversation is recordable - dialouts with CAPI are using early B3 - setting immediate=yes in capi.conf to prevent bugs with some external PBX - restarting asterisk causes kernel modules too reload - updated LCR script to v1.12 beta - verbose logging into a file is available - CallerID won't be set to 'unknown' if no CallerID was set in reverse lookup script - fixed cryptical entries in CallerIDs - minimized CRC errors on Zap channels - updated bristuff to 0.3.0-PRE-1q asterisk 1.1.3 -> 1.1.4 2006/01/26 unstable =========================================== - fixed bug that ASTERISK_VBOX_PLAY_INSTRUCTIONS was ignored - fixed bug in SIP callback (host wasn't given) - updated to bristuff 0.3.0-PRE-1i asterisk 1.1.2 -> 1.1.3 2006/01/13 unstable =========================================== - added asterisk-addons - recompiled chan_unistim for i586 instead of i686 - webinterface: fixed bug that one empty external MSN is displayed - config parser: comment handling improved - internal DISA MSN is now added to internal MSNs in capi.conf - added error handling and validity check of vbox variables - vboxes answers the line after the correct time - added language settings in all extensions - added callback - setting Zap default context to 'defaut' instead of 'outgoing' - fixed bug with HFCS cards in TE mode that no incoming and outgoing calls were possible - fixed bug in menu creation - added advanced error messages for outgoing calls - fixed cryptical CallerIDName, when using LCR script for outgoing calls - added parameter ASTERISK_DIALPREFIX_SHOW_IAX - DIALPREFIX_SHOW_* are working correctly now - updated to bristuff 0.3.0-PRE-1f - updated to chan_capi-cm 0.6.2 asterisk 1.1.1 -> 1.1.2 2006/01/02 unstable =========================================== - using german dial, ring, busy and congestion tones - fixed bug in plugin interface - webinterface: fixed bug that causes some MSNs to displayed twice - fixed error message after installation - fixed error in vbox generation script - updated internal Asterisk version to 1.2.1 - using bristuff 0.3.0-PRE-1d - using chan_capi-cm 0.6.1 instead of junghanns.net chan_capi asterisk 1.1.0 -> 1.1.1 2005/12/27 unstable =========================================== - fixed bug that vboxes aren't written to extensions.conf if they were defined in ASTERISK_VBOX_n_* - webinterface: fixed error message when displaying external MSNs asterisk 1.0.0 -> 1.1.0 2005/12/24 unstable =========================================== - inverse search: treating numbers without leading 0 but local area code correctly - restructured PHP scripts for configuration files creation - added support of FXO and FXS interfaces to connect analogue phones directly to the computer - added an interactive voice response menu - added IAX support - changed meaning of vboxes of internal phones (ASTERISK_PHONES_n_VBOX_*) see documentation - vboxes for external numbers may configured seperatly to achieve higher flexibility - added dynamic redirecting of incoming calls - renamed ASTERISK_CAPI_EXTRA_* to ASTERISK_MSN_* - added webinterface asterisk 0.0.18 -> 1.0.0 2005/10/18 stable =========================================== - changed status to stable asterisk 0.0.17 -> 0.0.18 2005/09/24 testing =========================================== - fixed bug when calling over CAPI by SIP client: CallerID wasn't set correctly asterisk 0.0.16 -> 0.0.17 2005/09/04 testing =========================================== - added plugin menu - fixed bug that callerID wasn't set correctly when using LCR over PSTN asterisk 0.0.15 -> 0.0.16 2005/08/28 unstable =========================================== - LCR configuration file will be updated, if the asterisk package is updated, too - fixed some bugs with ISDN-PtP connections - fallback is possible over HFCS cards - improved configuration generation with HFCS cards connected to the telco - suppressing LCR cronjob output - plays the busy tones if a number is busy and not the error announcement (wrong number dialed) - added LCR in the callthroughsystems and if falling back - adding a 0 to the calerid of incoming SIP connection if not given asterisk 0.0.14 -> 0.0.15 2005/08/24 unstable =========================================== - fixed bug that causes recorded vbox messages not to play - patched asterisk binary for full german locaization at the vboxes - fixed bug that vbox messages could only played at the vbox' phone - fixed bug in fallback, which didn't work asterisk 0.0.13 -> 0.0.14 2005/08/24 unstable =========================================== - fixed bug that no boxes answered the line - changed vbox fileformat to wav - fixed bug in inverse search script, that causes SIP numbers not to resolve nor to transmit asterisk 0.0.12 -> 0.0.13 2005/08/20 unstable =========================================== - updated bristuff to 0.2.0-RC8n - starting asterisk with nice -n -20 - localized voicebox menu - prevent loading of chan_oss.so - added automated LCR with telefon-sparbuch.de - added support for cisco IP phones (skinny) - added support for Nortel IP phones (unistim) - chmodding sip.conf to 600 - support of multiple HFC-S cards - support of the following junghanns.net HFC-S cards: singleE1, doubleE1, quadBRI - support of PtP connection (Anlagenanschluss) - a HFC-S card can be used (beside the Fritz!Card) to connect to an external S0 bus such as the PSTN or to an internal S0 bus of a telephone system - playing congestion after the call ends - fixed some bugs with internal calls redirected to a vbox (timeout problem) - changed vbox fileformat to wav49 asterisk 0.0.11 -> 0.0.12 2005/07/02 unstable =========================================== - set canreinvite for SIP clients to "no" - removed obsolete variable ASTERISK_CALLTHROUGH_n_PIN - alowing empty values for ASTERISK_CALLTHROUGH_n_ALLOW_m_PIN now - CAPI callerID for fallback calls will be set - SIP clients are also allowed if ASTERISK_SIP_N is set to 0 - added error announcement, when trying to call through SIP but ASTERISK_PHONES_n_OUTGOING_SIP is empty - added error announcement, when trying to call through CAPI but ASTERISK_PHONES_n_OUTGOING_MSN is empty - improved accesscontrol on callthrough - updated internal asterisk version to 1.0.9 - fixed bug in ASTERISK_CAPI_EXTRA_n_REDIRECT_* asterisk 0.0.10 -> 0.0.11 2005/06/25 unstable =========================================== - changes in asterisk.php for chan_capi 4PRE-1 asterisk 0.0.9 -> 0.0.10 2005/06/24 unstable =========================================== - updated internal Asterisk to version 1.0.8 due to stack overflow in the TAPI-Interface - fixed some bugs in the inverse search script, added some error handling asterisk 0.0.8 -> 0.0.9 2005/05/29 unstable =========================================== - fixed a bug in area code auto-completetion if a SIP dialprefix is set - improved support of T-Online SIP proxy asterisk 0.0.7 -> 0.0.8 2005/05/28 unstable =========================================== - fallback to other sip accounts or CAPI, if an outgoing SIP call fails (ASTERISK_SIP_n_FALLBACK) - preselection for outgoing CAPI calls (ASTERISK_CAPI_PRESELECTION) - now it's configurable, wether the dialprefix is shown or not in the incoming caller ID (ASTERISK_DIALPREFIX_SHOW_*) - setting area code in front of the phone number when dialing out via SIP and no leading 0 is given - writing and getting results by dasoertliche.de from/to /public/phonelist.txt - dialprefix won't be added anymore if the callerid is not transmitted - added some files to the deinstall script - added plugin interface - updated binaries to bristuff-0.2.0-RC8e with florz patch asterisk 0.0.6 -> 0.0.7 2005/05/05 unstable =========================================== - changed order of SetCIDName and SetCIDNum when calling an internal phone - setting dialprefix after calling the AGI-Script (invers search) for incoming calls - now Asterisk reacts only to the CAPI MSNs defined in the configuration - added var ASTERISK_ADVANCED_ERROR_MSG - creating spool directory during the installation - expanded check.d-files for support of Eisfair Configuration Editor (ECE) - suppressing output of depmod during installation - fixed bug in internal extensions configuration generation (doesn't work, if OUTGOING_MSN = MSN) - not playing congestion tones on dialouts asterisk 0.0.5 -> 0.0.6 2005/05/01 unstable =========================================== - dialprefix is part of the incoming CallerID displayed on the phone - fixed some bugs in the call redirection - installation won't be aborted if the unloading of the zap* modules fails - added var ASTERISK_SIP_n_CALLERID - added var ASTERISK_SIP_n_REDIRECT for enabling or disabling redirection of calls - added var ASTERISK_CAPI_EXTRA_n_REDIRECT for enabling or disabling redirection of calls - added var ASTERISK_VBOX_PLAY_INSTRUCTION - added support of web.de SIP proxy - added support of T-Online SIP proxy - added new error handling for invalid extensions - after finishing a call, congestion tones will be played asterisk 0.0.4 -> 0.0.5 2005/04/01 unstable =========================================== - incoming SIP-Calls setCallerID -> setCIDNum - check.d: ASTERISK_PHONES_n_MSN beside NUMERIC, * and # is allowed - setLanguage is now used at the VBox extensions - Asterisk provides a dialtone if DISA is used (Call attribute "r") - removed non-functional variable CALLERIDS in the callthrough section - the internal extentions haven't got a dial command if a VBox wasn't active - each codec in the sip.conf in the general section has it's own line now - renamed ASTERISK_CAPI_EXEC_* to ASTERISK_CAPI_EXTRA_* - added call forwarding (ASTERISK_CAPI_EXTRA_n_REDIRECT_*, ASTERISK_SIP_n_REDIRECT_*) - added ASTERISK_RTP_PORTS - Added option "s" to the Voicemail command asterisk 0.0.3 -> 0.0.4 2005/03/28 unstable =========================================== - made chan_capi work with i586 asterisk 0.0.2 -> 0.0.3 2005/03/27 unstable =========================================== - added prilocaldialplan=local in zapata.conf for removing the leading 0 in the caller ID - fixed bug at outgoing CAPI DISA calls - fixed bug at calculation of VBox timeouts - chan_capi.so is also compiled against i586 now - using VBox is also possible for incoming CAPI calls now - updated to bristuff-0.2.0-rc7k asterisk 0.0.1 -> 0.0.2 2005/02/05 unstable =========================================== - added TAPI support - added callthrough support - added invers search for telephone numbers - usable without HFC-S card - usable without SIP account - changed programing language in /var/install/config.d/* to PHP - compile target is now i586 - updated to bristuff-0.2.0-rc5 asterisk 0.0.0 -> 0.0.1 2004/12/24 unstable =========================================== - initial version