I set up Asterisk Certified and basically configured it according to this german how to: [Kurs] Wir konfigurieren uns einen Asterisk | IP Phone Forum
I set up an echo in extensions.conf like this
[echotest]
exten => 80,1,answer
exten => 80,2,wait,2
exten => 80,3,Playback(demo-echotest)
exten => 80,4,echo
exten => 80,5,Playback(demo-echodone)
exten => 80,6,hangup
and set up the 80 as a phone number to call in sip.conf
[80]
callerid=echotest <80>
domain=;INTERNAL IP OF THE SERVER
user=80
host=dynamic
secret= ; ************
nat=yes
type=friend
canreinvite=no
If I now call 80 through another configured phone number (i use phonerlite software) the call can be established according to phonerlite (every call can be esablished, even to phones not configured in sip.conf) - but theres silence at the other end, it should be at least the demo-echotest soundfile…
I get
ERROR[8321]: ari/config.c:296 process_config: No configured users for ARI
in the CLI after issuing a reload of the configuration… But I don’t know if that has anything to do with working calls…
If you need more, I can add some “asterisk -rvvv” → “reload” output or some other config files, if that helps.
For now I run Asterisk as a VM inside VMware Player, does the server need to have a sound card installed to play that sounds even through sip calls?
Edit: Reordered extensions.conf, It seems that I can call my Echo now, but I get the classic sound when the other place is busy…
in asterisk verbose 10 I get:
WARNING[1426][C-00000019]: pbx.c:4912 pbx_extension_helper: No application ‘wait,1’ for extension (default, 80, 2)
== Spawn extension (default, 80, 2) exited non-zero on ‘SIP/30-0000000a’