Hello Guys,
I downloaded Asterisk version 20.8.1 and ran it on a VirtualBox Ubuntu 24.04 LTS. I followed the installation guide from this LINK Afterward, I followed this tutorial, but tried to adapt it to my use case, meaning I used the chan_pjsip.so.
Here’s my current configuration:
ubuntu@ubuntu-VirtualBox:/etc/asterisk$ ls
asterisk.conf indications.conf pjsip_notify.conf sip.conf.default
cdr.conf logger.conf queues.conf voicemail.conf
cdr_custom.conf modules.conf README
confbridge.conf musiconhold.conf samples
extensions.conf pjsip.conf sip.conf
ubuntu@ubuntu-VirtualBox:/etc/asterisk$ cat sip.conf
[general]
udpbindaddr = 0.0.0.0:5060
context = dummy
disallow = all
allow = ulaw
alwaysauthreject = yes
allowguest = no
[zoiper]
type=friend
secret=#supersecret#
host=dynamic
qualify=yes
directmedia=no
context=from-internal
[xlite]
type=friend
secret=#supersecret#
host=dynamic
qualify=yes
directmedia=no
context=from-internal
ubuntu@ubuntu-VirtualBox:/etc/asterisk$ cat modules.conf
[modules]
autoload = yes
; This is a minimal module load. We are loading only the modules
; required for the Asterisk features used in the "Super Awesome
; Company" configuration.
; Applications
load = app_bridgewait.so
load = app_dial.so
load = app_playback.so
load = app_stack.so
load = app_verbose.so
load = app_voicemail.so
load = app_directory.so
load = app_confbridge.so
load = app_queue.so
; Bridging
load = bridge_builtin_features.so
load = bridge_builtin_interval_features.so
load = bridge_holding.so
load = bridge_native_rtp.so
load = bridge_simple.so
load = bridge_softmix.so
; Call Detail Records
load = cdr_custom.so
; Channel Drivers
load = chan_bridge_media.so
load = chan_pjsip.so
; Codecs
load = codec_gsm.so
load = codec_resample.so
load = codec_ulaw.so
load = codec_g722.so
; Formats
load = format_gsm.so
load = format_pcm.so
load = format_wav_gsm.so
load = format_wav.so
; Functions
load = func_callerid.so
load = func_cdr.so
load = func_pjsip_endpoint.so
load = func_sorcery.so
load = func_devstate.so
load = func_strings.so
; Core/PBX
load = pbx_config.so
; Resources
load = res_http_websocket.so
load = res_musiconhold.so
load = res_pjproject.so
load = res_pjsip_acl.so
load = res_pjsip_authenticator_digest.so
load = res_pjsip_caller_id.so
load = res_pjsip_dialog_info_body_generator.so
load = res_pjsip_diversion.so
load = res_pjsip_dtmf_info.so
load = res_pjsip_endpoint_identifier_anonymous.so
load = res_pjsip_endpoint_identifier_ip.so
load = res_pjsip_endpoint_identifier_user.so
load = res_pjsip_exten_state.so
load = res_pjsip_header_funcs.so
load = res_pjsip_logger.so
load = res_pjsip_messaging.so
load = res_pjsip_mwi_body_generator.so
load = res_pjsip_mwi.so
load = res_pjsip_nat.so
load = res_pjsip_notify.so
load = res_pjsip_one_touch_record_info.so
load = res_pjsip_outbound_authenticator_digest.so
load = res_pjsip_outbound_publish.so
load = res_pjsip_outbound_registration.so
load = res_pjsip_path.so
load = res_pjsip_pidf_body_generator.so
load = res_pjsip_pidf_digium_body_supplement.so
load = res_pjsip_pidf_eyebeam_body_supplement.so
load = res_pjsip_publish_asterisk.so
load = res_pjsip_pubsub.so
load = res_pjsip_refer.so
load = res_pjsip_registrar.so
load = res_pjsip_rfc3326.so
load = res_pjsip_sdp_rtp.so
load = res_pjsip_send_to_voicemail.so
load = res_pjsip_session.so
load = res_pjsip.so
load = res_pjsip_t38.so
load = res_pjsip_transport_websocket.so
load = res_pjsip_xpidf_body_generator.so
load = res_rtp_asterisk.so
load = res_sorcery_astdb.so
load = res_sorcery_config.so
load = res_sorcery_memory.so
load = res_sorcery_realtime.so
load = res_timing_timerfd.so
; Do not load res_hep and kin unless you are using HEP monitoring
; <http://sipcapture.org> in your network.
noload = res_hep.so
noload = res_hep_pjsip.so
noload = res_hep_rtcp.so
ubuntu@ubuntu-VirtualBox:/etc/asterisk$
What are the next steps to build a simple PBX where I can make outbound calls using Python and the Asterisk API? Can anyone help out, for a reasonable price or karma points?
Note: Any reading material is welcome, i am not a lazy person, just a noob 
Best,
Volkan
