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Das mit den Modulen ist gerade der "Trick" an einem statischen Build. Die Module werden hier nicht in einzelne Dateien ausgelagert. Statt dessen werden alle Module in das asterisk Binary hinein kompiliert. Die Module sind also alle schon mit dabei, werden nur nicht mehr dynamisch nachgeladen.
Hier ein Auszug meines aktuellen Builds.
Die Beispiel Conf-Dateien habe ich absichtlich nicht ins Archiv gepackt. Sonst überschreibt man sich beim Entpacken des Archivs nämlich schnell seine bereits existierenden Dateien. Ich kann die Beispieldateien aber auch hier nochmal als extra Archiv hochladen.
Alles was Capi angeht muss ich aber passen. Da kenne ich mich nicht aus, und habe auch keinerlei Möglichkeit, es irgendwie zu testen. Ich habe nämlich nur noch ausschließlich IP-Telefone. Diesem Thema müsste sich also jemand anderes widmen.
Hier ein Auszug meines aktuellen Builds.
Code:
root@freetz:/var/mod/root# asterisk -r
Asterisk 1.6.2.11, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.6.2.11 currently running on freetz (pid = 3530)
freetz*CLI> module show
Module Description Use Count
res_timing_pthread pthread Timing Interface 1
res_speech Generic Speech Recognition API 0
res_realtime Realtime Data Lookup/Rewrite 0
res_musiconhold Music On Hold Resource 0
res_limit Resource limits 0
res_curl cURL Resource Module 0
res_crypto Cryptographic Digital Signatures 0
res_convert File format conversion CLI command 0
res_config_curl Realtime Curl configuration 0
res_clioriginate Call origination and redirection from th 0
res_clialiases CLI Aliases 0
res_agi Asterisk Gateway Interface (AGI) 1
res_ael_share share-able code for AEL 0
pbx_spool Outgoing Spool Support 0
pbx_realtime Realtime Switch 0
pbx_loopback Loopback Switch 0
pbx_dundi Distributed Universal Number Discovery ( 0
pbx_config Text Extension Configuration 0
pbx_ael Asterisk Extension Language Compiler 0
func_volume Technology independent volume control 0
func_vmcount Indicator for whether a voice mailbox ha 0
func_version Get Asterisk Version/Build Info 0
func_uri URI encode/decode dialplan functions 0
func_timeout Channel timeout dialplan functions 0
func_sysinfo System information related functions 0
func_strings String handling dialplan functions 0
func_sprintf SPRINTF dialplan function 0
func_shell Returns the output of a shell command 0
func_sha1 SHA-1 computation dialplan function 0
func_realtime Read/Write/Store/Destroy values from a R 0
func_rand Random number dialplan function 0
func_module Checks if Asterisk module is loaded in m 0
func_md5 MD5 digest dialplan functions 0
func_math Mathematical dialplan function 0
func_logic Logical dialplan functions 0
func_lock Dialplan mutexes 0
func_iconv Charset conversions 0
func_groupcount Channel group dialplan functions 0
func_global Variable dialplan functions 0
func_extstate Gets an extension's state in the dialpla 0
func_env Environment/filesystem dialplan function 0
func_enum ENUM related dialplan functions 0
func_dialplan Dialplan Context/Extension/Priority Chec 0
func_dialgroup Dialgroup dialplan function 0
func_devstate Gets or sets a device state in the dialp 0
func_db Database (astdb) related dialplan functi 0
func_cut Cut out information from a string 0
func_curl Load external URL 0
func_config Asterisk configuration file variable acc 0
func_channel Channel information dialplan functions 0
func_cdr Call Detail Record (CDR) dialplan functi 0
func_callerid Caller ID related dialplan functions 0
func_blacklist Look up Caller*ID name/number from black 0
func_base64 base64 encode/decode dialplan functions 0
func_audiohookinherit Audiohook inheritance function 0
func_aes AES dialplan functions 0
format_wav_gsm Microsoft WAV format (Proprietary GSM) 0
format_wav Microsoft WAV format (8000Hz Signed Line 0
format_vox Dialogic VOX (ADPCM) File Format 0
format_sln Raw Signed Linear Audio support (SLN) 0
format_sln16 Raw Signed Linear 16KHz Audio support (S 0
format_siren7 ITU G.722.1 (Siren7, licensed from Polyc 0
format_siren14 ITU G.722.1 Annex C (Siren14, licensed f 0
format_pcm Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0
format_jpeg jpeg (joint picture experts group) image 0
format_ilbc Raw iLBC data 0
format_h264 Raw H.264 data 0
format_h263 Raw H.263 data 0
format_gsm Raw GSM data 0
format_g729 Raw G729 data 0
format_g726 Raw G.726 (16/24/32/40kbps) data 0
format_g723 G.723.1 Simple Timestamp File Format 0
codec_ulaw mu-Law Coder/Decoder 0
codec_lpc10 LPC10 2.4kbps Coder/Decoder 0
codec_gsm GSM Coder/Decoder 0
codec_g726 ITU G.726-32kbps G726 Transcoder 0
codec_g722 ITU G.722-64kbps G722 Transcoder 0
codec_a_mu A-law and Mulaw direct Coder/Decoder 0
codec_alaw A-law Coder/Decoder 0
codec_adpcm Adaptive Differential PCM Coder/Decoder 0
chan_sip Session Initiation Protocol (SIP) 0
chan_local Local Proxy Channel (Note: used internal 0
chan_iax2 Inter Asterisk eXchange (Ver 2) 0
chan_datacard Datacard Channel Driver 0
chan_bridge Bridge Interaction Channel 0
cdr_custom Customizable Comma Separated Values CDR 0
cdr_csv Comma Separated Values CDR Backend 0
bridge_softmix Multi-party software based channel mixin 0
bridge_simple Simple two channel bridging module 0
bridge_multiplexed Multiplexed two channel bridging module 0
bridge_builtin_features Built in bridging features 1
app_while While Loops and Conditional Execution 0
app_waituntil Wait until specified time 0
app_waitforsilence Wait For Silence 0
app_waitforring Waits until first ring after time 0
app_voicemail Comedian Mail (Voicemail System) 0
app_userevent Custom User Event Application 0
app_url Send URL Applications 0
app_transfer Transfers a caller to another extension 0
app_talkdetect Playback with Talk Detection 0
app_system Generic System() application 0
app_stack Dialplan subroutines (Gosub, Return, etc 0
app_softhangup Hangs up the requested channel 0
app_sendtext Send Text Applications 0
app_senddtmf Send DTMF digits Application 0
app_sayunixtime Say time 0
app_record Trivial Record Application 0
app_readfile Stores output of file into a variable 0
app_readexten Read and evaluate extension validity 0
app_read Read Variable Application 0
app_privacy Require phone number to be entered, if n 0
app_playtones Playtones Application 0
app_playback Sound File Playback Application 0
app_parkandannounce Call Parking and Announce Application 0
app_originate Originate call 0
app_mixmonitor Mixed Audio Monitoring Application 0
app_macro Extension Macros 0
app_followme Find-Me/Follow-Me Application 0
app_exec Executes dialplan applications 0
app_echo Simple Echo Application 0
app_dumpchan Dump Info About The Calling Channel 0
app_disa DISA (Direct Inward System Access) Appli 0
app_directory Extension Directory 0
app_directed_pickup Directed Call Pickup Application 0
app_dictate Virtual Dictation Machine 0
app_dial Dialing Application 0
app_db Database Access Functions 0
app_controlplayback Control Playback Application 0
app_confbridge Conference Bridge Application 0
app_chanspy Listen to the audio of an active channel 0
app_channelredirect Redirects a given channel to a dialplan 0
app_chanisavail Check channel availability 0
app_cdr Tell Asterisk to not maintain a CDR for 0
app_authenticate Authentication Application 0
134 modules loadedDie Beispiel Conf-Dateien habe ich absichtlich nicht ins Archiv gepackt. Sonst überschreibt man sich beim Entpacken des Archivs nämlich schnell seine bereits existierenden Dateien. Ich kann die Beispieldateien aber auch hier nochmal als extra Archiv hochladen.
Alles was Capi angeht muss ich aber passen. Da kenne ich mich nicht aus, und habe auch keinerlei Möglichkeit, es irgendwie zu testen. Ich habe nämlich nur noch ausschließlich IP-Telefone. Diesem Thema müsste sich also jemand anderes widmen.
