Hi Pilovis!
I've installed the Asterisk 11 packages based on your tutorial on my asus router (AC56U), the Huawei dongle E169 is connected to the USB port 3.0.
Sms sending is working.
Voip calls are working between SIP clients.
From external calling the dongle I got a busy tone and on the logs I don't see anything
From sip client to external call (603/Declined - on the client), on the asterisk cli :
RT-AC56U-8B58*CLI>
== Using SIP RTP CoS mark 5
-- Executing [XXXXXXXXXX@from-sip:1] Log("SIP/sanyi-00000002", "NOTICE, Dialing out from "" <sanyi> to XXXXXXXXXX through Dongle0 Provider") in new stack
[Aug 3 13:31:40] NOTICE[4436][C-00000002]: Ext. 0770208650:1 @ from-sip: Dialing out from "" <sanyi> to 0770208650 through Dongle0 Provider
-- Executing [XXXXXXXXXX@from-sip:2] Dial("SIP/sanyi-00000002", "dongle/dongle0/XXXXXXXXXX,60") in new stack
[Aug 3 13:31:40] WARNING[4436][C-00000002]: channel.c:180 channel_request: [dongle0] Request to call on device which can not make call at this moment
[Aug 3 13:31:40] WARNING[4436][C-00000002]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'dongle' (cause 44 - Requested channel not available)
== Everyone is busy/congested at this time (1:0/0/1)
[Aug 3 13:31:40] WARNING[4436][C-00000002]: pbx.c:4926 pbx_extension_helper: No application 'Playtones' for extension (from-sip, XXXXXXXXXX, 3)
== Spawn extension (from-sip,XXXXXXXXXX, 3) exited non-zero on 'SIP/sanyi-00000002'
dongle show device state dongle0
-------------- Status -------------
Device : dongle0
State : Dialing
Audio : /dev/ttyUSB0
Data : /dev/ttyUSB1
Voice : Yes
SMS : Yes
Manufacturer : huawei
Model : E169
Firmware : 11.315.05.00.00
IMEI : 359638015545142
IMSI : 226050080737498
GSM Registration Status : Registered, home network
RSSI : 6, -101 dBm
Mode : No Service
Submode : No service
Provider Name : Digi.Mobil
Location area code : 2C1
Cell ID : 3DC3
Subscriber Number : +4YYYYYYYYYY
SMS Service Center : +40770000050
Use UCS-2 encoding : Yes
USSD use 7 bit encoding : No
USSD use UCS-2 decoding : Yes
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 : 1
Active : 0
Held : 0
Dialing : 0
Alerting : 0
Incoming : 0
Waiting : 0
Releasing : 0
Initializing : 1
RT-AC56U-8B58*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
bobby/bobby (Unspecified) D Auto (No) No 0 UNKNOWN
sanyi/sanyi 192.168.32.175 D Auto (No) No 58554 OK (238 ms)
2 sip peers [Monitored: 1 online, 1 offline Unmonitored: 0 online, 0 offline]
extensions.conf:
[from-sip]
exten => 100,1,Dial(SIP/sanyi,30)
exten => 100,2,Hangup()
exten => 101,1,Dial(SIP/bobby,30)
exten => 100,2,Hangup()
exten => _X.,1,Log(NOTICE, Dialing out from ${CALLERID(all)} to ${EXTEN} through Dongle0 Provider)
exten => _X.,n,Dial(dongle/dongle0/${EXTEN},60)
exten => _X.,n,Playtones(congestion)
exten => _X.,n,Hangup()
[from-dongle]
exten => +4YYYYYYYYYY,1,Log(NOTICE, Incoming call from ${CALLERID(all)})
exten => +4YYYYYYYYYY,n,Dial(SIP/sanyi)
exten => +4YYYYYYYYYY,n,Hangup()
dongle.conf:
.......
[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=from-dongle ; context for incoming calls
group=1 ; 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
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'
exten=+4YYYYYYYYYY ; 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:
[dongle0]
audio=/dev/ttyUSB0 ; tty port for audio connection; no default value
data=/dev/ttyUSB1 ; tty port for AT commands; no default value
; 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=359638015545142
imsi=226050080737498
sip.conf:
......
[sanyi]
type=friend
username=sanyi
secret=******
host=dynamic
context=from-sip
[bobby]
type=friend
username=bobby
secret=******
host=dynamic
context=from-sip
Any ideas with the busy issue?
Could be locked/disabled the Voice feature, even if the we see "Voice: Yes" on the dongle status?
The AT^CVOICE=? -> responds: AT^CVOICE:(0) - OK
A DC-UNCLOCKER from windows says VOICE disabled.
I am confused a little
.