I'd really like to make and perfect it myself if that's possible. There's so many things I want to trim out - including the generic comment lines in the /etc/asterisk config files, and a bunch of those configs. I don't want to compile any of the following libraries which I have a script that removes as follows:
cd /usr/lib/asterisk/modules
rm app_a* app_c* app_db.so app_directory.so app_disa.so app_enumlookup.so app_eval.so app_exec.so app_f* app_g* app_i* app_l*
rm app_parkandannounce.so app_privacy.so app_q* app_r* app_se* app_sms.so app_softhangup.so app_striplsd.so
rm app_substring.so app_system.so app_talkdetect.so app_test.so app_txtcidname.so app_u* app_w* cdr_*
rm chan_agent.so chan_mgcp.so chan_skinny.so codec_a* codec_g726.so codec_ilbc.so codec_lpc10.so
rm format_g72* format_h263.so format_ilbc.so format_pcm_alaw.so format_sln.so format_vox.so
rm pbx_spool.so pbx_wilcalu.so res_agi.so res_monitor.so
None of those are necessary for connection to an IAX termination service such as diamondcard.us or teliax.com. Without those files asterisk runs fine, although it requires a pretty restrictive modules.conf, and the removal of several configs in /etc/asterisk. My current revision of modules.conf looks like this:
[modules]
autoload=no
load=res_features.so ; Call Parking Resource
load=chan_iax2.so ; Inter Asterisk eXchange (Ver 2)
load=chan_local.so ; Local Proxy Channel
load=chan_sip.so ; Session Initiation Protocol (SIP)
load=codec_gsm.so ; GSM/PCM16 (signed linear) Codec Translat
load=codec_ulaw.so ; Mu-law Coder/Decoder
load=format_gsm.so ; Raw GSM data
load=format_pcm.so ; Raw uLaw 8khz Audio support (PCM)
load=format_wav.so ; Microsoft WAV format (8000hz Signed Line
load=format_wav_gsm.so ; Microsoft WAV format (Proprietary GSM)
load=app_dial.so ; Dialing Application
load=app_echo.so ; Simple Echo Application
load=app_macro.so ; Extension Macros
load=app_milliwatt.so ; Digital Milliwatt (mu-law) Test Applicat
load=app_playback.so ; Trivial Playback Application
load=app_sayunixtime.so ; Say time
load=app_transfer.so ; Transfer
load=app_verbose.so ; Send verbose output
load=pbx_config.so ; Text Extension Configuration
[global]
On top of this, as mentioned earlier, I have the number of files required in /etc/asterisk down to 10, give or take a couple.
I believe there is still a file or two that needs to be put in to solve the hook flash problem. This is something I'm prepared to spend a lot of time with if someone can bring me into the loop with the SDK error. I sincerely appreciate the offer to make a mini-package, but, I believe I'm going to want to create a lot of them for testing, and I wouldn't want to ask you to spend that much time on it.
Tom