Hello Community!
I’ve chosen Asterisk for my home automation project. Hope that my final target will be achievable but for now I’ve got stuck on probably simple thing… I am not able to react on any call incoming to the dongle - the line is incessantly busy. I would like to hear just anything like, let’s say, SayDigits(23456789) or any other “hello world”. It should be easy, right? 
My current conf:
Ubuntu 17.10 + MSISDN + HUAWEI E173u-2 + Asterisk + asterisk-chan-dongle
MSISDN on dongle is, let’s say +48xxxyyyzzz, (will use this below)
Assuming that dongle is correctly installed because:
Loaded chan_dongle.so => (Huawei 3G Dongle Channel Driver)
-- [dongle0] IMEI 8XXXXXXXX found on data_tty=/dev/ttyUSB2 audio_tty=/dev/ttyUSB1
-- [dongle0] Trying to connect on /dev/ttyUSB2...
-- [dongle0] Dongle has connected, initializing...
-- [dongle0] Dongle initialized and ready
plus: I can receive and send SMS. Dongle is unlocked, voice is enabled.
My dongle properties:
dongle show device state dongle0
-------------- Status -------------
Device : dongle0
State : Free
Audio : /dev/ttyUSB1
Data : /dev/ttyUSB2
Voice : Yes
SMS : Yes
Manufacturer : huawei
Model : E173
Firmware : 11.126.85.02.52
IMEI : 8--------------0
IMSI : 2---------------0
GSM Registration Status : Registered, home network
RSSI : 10, -93 dBm
Mode : No Service
Submode : No service
Provider Name : Era
Location area code : XXX
Cell ID : XXX
Subscriber Number : +48XXXXXXXX
SMS Service Center : +48XXXXXXXX
Use UCS-2 encoding : Yes
USSD use 7 bit encoding : Yes
USSD use UCS-2 decoding : No
Tasks in queue : 0
Commands in queue : 0
Call Waiting : Disabled
Current device state : start
Desired device state : start
When change state : now
Calls/Channels : 0
Active : 0
Held : 0
Dialing : 0
Alerting : 0
Incoming : 0
Waiting : 0
Releasing : 0
Initializing : 0
My Asterisk version is:
PBX Core settings
-----------------
Version: 15.2.2
Build Options: BUILD_NATIVE, OPTIONAL_API
Maximum calls: Not set
Maximum open file handles: 1024
Root console verbosity: 0
Current console verbosity: 5
Debug level: 6
Maximum load average: 0.000000
Minimum free memory: 0 MB
Startup time: 20:31:55
Last reload time: 20:38:50
System: Linux/4.13.0-36-generic built by root on x86_64 2018-03-06 21:49:35 UTC
System name:
Entity ID: 4:::::f
PBX UUID: 2-----8
Default language: en
Language prefix: Enabled
User name and group: /
Executable includes: Disabled
Transcode via SLIN: Enabled
Transmit silence during rec: Disabled
Generic PLC: Disabled
Min DTMF duration:: 80
Cache media frames: Enabled
RTP use dynamic payloads: 1
RTP dynamic payload types: 35-63,96-127
* Subsystems
-------------
Manager (AMI): Disabled
Web Manager (AMI/HTTP): Disabled
Call data records: Enabled
Realtime Architecture (ARA): Disabled
* Directories
-------------
Configuration file: /etc/asterisk/asterisk.conf
Configuration directory: /etc/asterisk
Module directory: /usr/lib/asterisk/modules
Spool directory: /var/spool/asterisk
Log directory: /var/log/asterisk
Run/Sockets directory: /var/run/asterisk
PID file: /var/run/asterisk/asterisk.pid
VarLib directory: /var/lib/asterisk
Data directory: /var/lib/asterisk
ASTDB: /var/lib/asterisk/astdb
IAX2 Keys directory: /var/lib/asterisk/keys
AGI Scripts directory: /var/lib/asterisk/agi-bin
Tried to configure /etc/asterisk/dongle.conf as below:
[general]
interval=3 ; Number of seconds between trying to connect to devices
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
;jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; Dongle channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The Dongle channel can't accept jitter,
; thus an enabled jitterbuffer on the receive Dongle side will always
; be used if the sending side can create jitter.
;jbforce = no ; Forces the use of a jitterbuffer on the receive side of a Dongle
; channel. Defaults to "no".
;jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
;jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
;jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a Dongle
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
;jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
; The option represents the number of milliseconds by which the new jitter buffer
; will pad its size. the default is 40, so without modification, the new
; jitter buffer will set its size to the jitter value plus 40 milliseconds.
; increasing this value may help if your network normally has low jitter,
; but occasionally has spikes.
;jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
[defaults]
; now you can set here any not required device settings as template
; sure you can overwrite in any [device] section this default values
context=helloWorld ; context for incoming calls
group=0 ; calling group
rxgain=0 ; increase the incoming volume; may be negative
txgain=0 ; increase the outgoint volume; may be negative
autodeletesms=yes ; auto delete incoming sms
resetdongle=yes ; reset dongle during initialization with ATZ command
u2diag=-1 ; set ^U2DIAG parameter on device (0 = disable everything except modem function) ; -1 not use ^U2DIAG command
usecallingpres=yes ; use the caller ID presentation or not
callingpres=allowed_passed_screen ; set caller ID presentation by default use default network settings
disablesms=no ; disable of SMS reading from device when received
; chan_dongle has currently a bug with SMS reception. When a SMS gets in during a
; call chan_dongle might crash. Enable this option to disable sms reception.
; default = no
language=en ; set channel default language
smsaspdu=yes ; if 'yes' send SMS in PDU mode, feature implementation incomplete and we strongly recommend say 'yes'
mindtmfgap=45 ; minimal interval from end of previews DTMF from begining of next in ms
mindtmfduration=80 ; minimal DTMF tone duration in ms
mindtmfinterval=200 ; minimal interval between ends of DTMF of same digits in ms
callwaiting=auto ; if 'yes' allow incoming calls waiting; by default use network settings
; if 'no' waiting calls just ignored
disable=no ; OBSOLETED by initstate: if 'yes' no load this device and just ignore this section
initstate=start ; specified initial state of device, must be one of 'stop' 'start' 'remote'
; 'remove' same as 'disable=yes'
;exten= ; exten for start incoming calls, only in case of Subscriber Number not available!, also set to CALLERID(ndid)
dtmf=relax ; control of incoming DTMF detection, possible values:
; off - off DTMF tones detection, voice data passed to asterisk unaltered
; use this value for gateways or if not use DTMF for AVR or inside dialplan
; inband - do DTMF tones detection
; relax - like inband but with relaxdtmf option
; default is 'relax' by compatibility reason
; dongle required settings
[dongle0]
audio=/dev/ttyUSB1 ; tty port for audio connection
data=/dev/ttyUSB2 ; tty port for AT commands
; or you can omit both audio and data together and use imei=123456789012345 and/or imsi=123456789012345
; imei and imsi must contain exactly 15 digits !
; imei/imsi discovery is available on Linux only
imei=XXXXXXXXXXXXXXXX
;imsi=123456789012345
; if audio and data set together with imei and/or imsi audio and data has precedence
; you can use both imei and imsi together in this case exact match by imei and imsi required
one of dozens /etc/asterisk/extensions.conf variants is as below:
[helloWorld]
exten => _X.,1,Answer()
exten => _X.,2,SayDigits(23456789)
exten => _X.,3,Hangup()
and that’s all of my configurations.
My assumption is that in example above ANY incoming call to dongle (+48xxxyyyzzz) will be taken, I will hear a voice which reads the numbers and then call will be hanged up. Am I mistaken? But I hear always busy line…
tried to do some debugging:
asterisk -rvvvvvvvvvddddddddddddd
but when I am calling to dongle nothing appears on the screen.
Could you, please, help me understand what is missed? Do you know any documentation that could help me? How to move forward?
Thanks!
