Moin
der Thread ist zwar schon alt, aber vielleicht findet sich ja ein asterisk Profi der mir helfen kann.
Ich habe den iobroker asterisk adapter auf einem anderen host wie den asterisk server und alles entsprechend https://github.com/schmupu/ioBroker.asterisk/blob/master/docs/SSH_DE.md eingerichtet.
Der iobroker meldet sich korrekt am remote asterisk an und schickt auch via ssh/scp die generierten audio files rüber.
Leider scheint dort irgend was mit er Konfiguration schief zu laufen.
Der asterisk adapter im iobroker meldet dann "Originate failed:
Error while dialing (1). Error: {"response":"Error","actionid":"ee1bb6cf-1fe7-80ae-252c-8ba913a0d052","message":"Originate failed"}, Result: {"parameter":{"callerid":"**620","telnr":"017643282951","text":"Hallo!","extension":"Smarthome","audiofile":"/tmp//audio_1234","delete":"delete"},"options":{"action":"originate","channel":"SIP/Smarthome/1234","context":"ael-ansage","exten":"1234","priority":1,"timeout":60000,"variable":{"repeat":5,"file":"/tmp//audio_1234","del":"delete"},"callerid":"**620"},"guid":"ee1bb6cf-1fe7-80ae-252c-8ba913a0d052","result":{"response":"Error","actionid":"ee1bb6cf-1fe7-80ae-252c-8ba913a0d052","message":"Originate failed"}}
Der asterisk sagt im debug mode dies:
asterisk -rvvvvv
Asterisk 18.24.3, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
Created by Mark Spencer <markster@digium.com>
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 18.24.3 currently running on mediau (pid = 8597)
mediau*CLI> core set debug 10
Core debug was OFF and is now 10.
== Manager 'manager' logged on from 10.0.0.3
[Sep 21 13:23:14] WARNING[8682]: channel.c:6332 request_channel: No channel type registered for 'SIP'
-- Executing [failed@ael-ansage:1] Answer("OutgoingSpoolFailed", "") in new stack
-- Executing [failed@ael-ansage:2] Wait("OutgoingSpoolFailed", "1") in new stack
-- Executing [failed@ael-ansage:3] Read("OutgoingSpoolFailed", "dtmf,/tmp//audio_1726924993523&beep,0,s,5,1") in new stack
-- Executing [failed@ael-ansage:4] GotoIf("OutgoingSpoolFailed", "0?5:6") in new stack
-- Goto (ael-ansage,failed,6)
-- Executing [failed@ael-ansage:6] NoOp("OutgoingSpoolFailed", "Finish if_ael-ansage_1") in new stack
-- Executing [failed@ael-ansage:7] Hangup("OutgoingSpoolFailed", "") in new stack
== Spawn extension (ael-ansage, failed, 7) exited non-zero on 'OutgoingSpoolFailed'
-- Executing [h@ael-ansage:1] GotoIf("OutgoingSpoolFailed", "1?2:4") in new stack
-- Goto (ael-ansage,h,2)
-- Executing [h@ael-ansage:2] NoOp("OutgoingSpoolFailed", "/bin/rm /tmp//audio_1726924993523.*") in new stack
-- Executing [h@ael-ansage:3] System("OutgoingSpoolFailed", "/bin/rm /tmp//audio_1726924993523.*") in new stack
-- Executing [h@ael-ansage:4] NoOp("OutgoingSpoolFailed", "Finish if_ael-ansage_2") in new stack
-- Executing [h@ael-ansage:5] SayDigits("OutgoingSpoolFailed", "") in new stack
-- Executing [h@ael-ansage:6] NoOp("OutgoingSpoolFailed", "Finish if_ael-ansage_1") in new stack
-- Executing [h@ael-ansage:7] Hangup("OutgoingSpoolFailed", "") in new stack
== Spawn extension (ael-ansage, h, 7) exited non-zero on 'OutgoingSpoolFailed'
ich habe keine Ahnung was der Asterisk Channel ist, der hier fehlt.
Meine sip.com
[general] ; Do not change
port = 5060 ; Do not change
bindaddr = 0.0.0.0 ; Do not change
context = default ; Do not change
subscribecontext = default ; Do not change
register => 62162162:pw****@10.0.0.1/1000 ; Username, Password and IP address of Fritzbox WLAN/LAN telephone
[62162162] ; Change to username of Fritzbox WLAN/LAN telephone
type = friend ; Do not change
username = 62162162 ; Change to username of Fritzbox WLAN/LAN telephone
host = 10.0.0.1 ; Change hostname / IP address of Fritzbox
secret = pw****** ; Change password of Fritzbox WLAN/LAN telephone
fromdomain = 10.0.0.1 ; Change hostname / IP address of Fritzbox
fromuser = 62162162 ; Change username of Fritzbox WLAN/LAN telephone
context = default
der 62162162 ist der username des SIP accounts in meine Fritzbox
extentions.ael ist die oben beschrieben datei:
context default {
1000 => {
Goto(ael-antwort,s,1);
}
}
context ael-ansage {
_. => {
Answer();
Wait(1);
Read(dtmf,${file}&beep,0,s,${repeat},1);
if ("${dtmf}" != "") {
SayDigits(${dtmf});
}
Hangup();
}
h => {
if ("${del}" = "delete") {
NoOp(/bin/rm ${file}.*);
System(/bin/rm ${file}.*);
}
}
}
context ael-antwort {
s => {
Answer();
Wait(1);
Set(repeat=5);
Read(dtmf,/tmp//asterisk_dtmf&beep,0,s,${repeat},1);
if ("${dtmf}" != "") {
SayDigits(${dtmf});
}
Hangup();
}
_. => {
Goto(ael-antwort,s,1);
}
}
Die extenstions.conf ist der standard den die 1000 definiert
...
exten => 1000,1,Goto(default,s,1)
...
Irgend wie scheint in der asterisk config was faul zu sein.
Kann jemand helfen?
Lieben Dank
Stefan
