My Asterisk adapter is working fine so far, but now i have a problem when i enter DTMF codes.
The codes don´t show up in the states of iobroker.
I create a Dial-out call to a number and playing an mp3 file. This works fine. But when i enter a DTMF Code the Code isn´t visible in the states of iobroker.
Here the command shell output when the call is established:
root@iobroker:/opt/iobroker/iobroker-data# asterisk -rvvvvvv
Asterisk 16.16.1~dfsg-1+deb11u1, Copyright (C) 1999 - 2018, Digium, Inc. 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 16.16.1~dfsg-1+deb11u1 currently running on iobroker (pid = 17817)
== Using SIP RTP CoS mark 5
-- Called 124516327/TARGET-NUMBER
0xffff80010500 -- Strict RTP learning after remote address set to: 192.168.178.1:7084
-- SIP/124516327-00000016 is making progress
0xffff80010500 -- Strict RTP switching to RTP target address 192.168.178.1:7084 as source
0xffff80010500 -- Strict RTP learning complete - Locking on source address 192.168.178.1:7084
-- SIP/124516327-00000016 answered
-- Executing [TARGET-NUMBER@ael-ansage:1] Answer("SIP/124516327-00000016", "") in new stack
-- Executing [TARGET-NUMBER@ael-ansage:2] Wait("SIP/124516327-00000016", "1") in new stack
-- Executing [TARGET-NUMBER@ael-ansage:3] Read("SIP/124516327-00000016", "dtmf,/tmp/audio&beep,0,s,5,1") in new stack
-- <SIP/124516327-00000016> Playing '/tmp/audio.gsm' (language 'en')
-- User entered '845'
-- Executing [TARGET-NUMBER@ael-ansage:4] GotoIf("SIP/124516327-00000016", "1?5:6") in new stack
-- Goto (ael-ansage,TARGET-NUMBER,5)
-- Executing [TARGET-NUMBER@ael-ansage:5] SayDigits("SIP/124516327-00000016", "845") in new stack
-- <SIP/124516327-00000016> Playing 'digits/8.gsm' (language 'en')
-- <SIP/124516327-00000016> Playing 'digits/4.gsm' (language 'en')
-- <SIP/124516327-00000016> Playing 'digits/5.gsm' (language 'en')
-- Executing [TARGET-NUMBER@ael-ansage:6] NoOp("SIP/124516327-00000016", "Finish if_ael-ansage_1") in new stack
-- Executing [TARGET-NUMBER@ael-ansage:7] Hangup("SIP/124516327-00000016", "") in new stack
== Spawn extension (ael-ansage, TARGET-NUMBER, 7) exited non-zero on 'SIP/124516327-00000016'
-- Executing [h@ael-ansage:1] GotoIf("SIP/124516327-00000016", "0?2:4") in new stack
-- Goto (ael-ansage,h,4)
-- Executing [h@ael-ansage:4] NoOp("SIP/124516327-00000016", "Finish if_ael-ansage_2") in new stack
-- Executing [h@ael-ansage:5] SayDigits("SIP/124516327-00000016", "845") in new stack
-- <SIP/124516327-00000016> Playing 'digits/8.gsm' (language 'en')
== Spawn extension (ael-ansage, h, 5) exited non-zero on 'SIP/124516327-00000016'
iobroker*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
root@iobroker:/opt/iobroker/iobroker-data#
Can someone tell me how to solve the problem? English or German is fine. Thanks.
My Asterisk adapter is working fine so far, but now i have a problem when i enter DTMF codes. The codes don´t show up in the states of iobroker.
I create a Dial-out call to a number and playing an mp3 file. This works fine. But when i enter a DTMF Code the Code isn´t visible in the states of iobroker.
Here the command shell output when the call is established:
root@iobroker:/opt/iobroker/iobroker-data# asterisk -rvvvvvv Asterisk 16.16.1~dfsg-1+deb11u1, Copyright (C) 1999 - 2018, Digium, Inc. 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 16.16.1~dfsg-1+deb11u1 currently running on iobroker (pid = 17817) == Using SIP RTP CoS mark 5 -- Called 124516327/TARGET-NUMBER
Can someone tell me how to solve the problem? English or German is fine. Thanks.