VasOleMil
commented
1 year ago Updated modem firmware to the latest 11.314.17.00.00->11.315.05.00.00, still no audio at GSM->SIP in Asterisk:
CLI> dongle show devices
ID Group State RSSI Mode Submode Provider Name Model Firmware IMEI IMSI Number
vodafone 1 Free 16 5 4 Vodafone UA E169 11.315.05.00.00 35963801387**** 25501515034**** Unknown Using simple configs for test purpase;
dongle.conf
[general]
interval=15
smsdb=/var/db/asterisk/smsdb
csmsttl=600
[defaults]
context=default
group=0
rxgain=0
txgain=0
autodeletesms=yes
resetdongle=yes
u2diag=-1
usecallingpres=yes
callingpres=allowed_passed_screen
disablesms=no
language=en
mindtmfgap=45
mindtmfduration=80
mindtmfinterval=200
callwaiting=auto
disable=no
initstate=start
exten=+38095*******
dtmf=relax
[vodafone]
context=vodafone-incoming
exten=+38095*******
audio=/dev/cuaU0.1
data=/dev/cuaU0.2
imei=35963801387****
imsi=25501515034****
group=1
rxgain=0
txgain=0
resetdatacard=yes
u2diag=0
usecallingpres=yes
callingpres=allowed_passed_screen
disablesms=nopjsip.conf
;===============TRANSPORT================================================================
[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.0.0/24
external_media_address=176.36.XXX.XXX
external_signaling_address=176.36.XXX.XXX
;===============ENDPOINT TEMPLATES=======================================================
[endpoint-basic](!)
type=endpoint
context=internal
disallow=all
allow=ulaw
[auth-userpass](!)
type=auth
auth_type=userpass
[aor-single-reg](!)
type=aor
max_contacts=1
remove_existing=yes
remove_unavailable=yes
;===============EXTENSION 6000
[6000](endpoint-basic)
auth=auth6000
aors=6000
[auth6000](auth-userpass)
password=6000
username=6000
[6000](aor-single-reg)
;===============EXTENSION 6001
[6001](endpoint-basic)
auth=auth6001
aors=6001
[auth6001](auth-userpass)
password=6001
username=6001
[6001](aor-single-reg)
extensions.conf
[vodafone-incoming-sms]
exten => sms,1,Noop(Incoming SMS from ${CALLERID(num)} ${BASE64_DECODE(${SMS_BASE64})})
exten => sms,n,System(echo '${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} - ${DONGLENAME} - ${CALLERID(num)}: ${BASE64_DECODE(${SMS_BASE64})}' >> /var/log/asterisk/sms.log)
exten => sms,n,Hangup()
[vodafone-incoming-ussd]
exten => ussd,1,Noop(Incoming USSD: ${BASE64_DECODE(${USSD_BASE64})})
exten => ussd,n,System(echo '${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} - ${DONGLENAME} - ${CALLERID(num)}: ${BASE64_DECODE(${USSD_BASE64})}' >> /var/log/asterisk/ussd.log)
exten => ussd,n,Hangup()
[vodafone-incoming]
include => vodafone-incoming-sms
include => vodafone-incoming-ussd
;redirect to internal number
exten => +38095*******,1,Dial(PJSIP/6000,60)
;play hellow world
;exten => +38095*******,1,Answer(500)
;exten => +38095*******,n,Playback(hello-world)
;exten => h,n,Hangup()
[internal]
exten = _6XXX,1,Dial(PJSIP/${EXTEN})
;external number call filter
exten => _38050XXXXXXX,1,Dial(Dongle/vodafone/+${EXTEN},240,tT)
exten => _38066XXXXXXX,1,Dial(Dongle/vodafone/+${EXTEN},240,tT)
exten => _38095XXXXXXX,1,Dial(Dongle/vodafone/+${EXTEN},240,tT)
;gsm provider call
exten => 111,1,Dial(Dongle/vodafone/${EXTEN},240,tT)
wdoekes
No audio from GSM to SIP, modem is working well with Windows 10 and provider program on the same machine, audio present in both directions regardless of caller. Asterisk installed from ports, SIP<->SIP connection works fine.
making sequence:
CLI check:
System check:
The only warning in console, when calling GSM->SIP at the end of connection:
WARNING[102589][C-00000002]: channel.c:1079 channel_indicate: [Dongle/vodafone-0100000001] Don't know how to indicate condition 22