Hi everyone,
I have problem with configure inbound calls from chan_dongle.
Sets witch GUI
TRUNK
Trunk Name = GSM
Custom Dial String = dongle/dongle0/$OUTNUM$
Outbound CallerID = 720744425
Inbound Routes
Description = GSM_IN
DID Number = 720744425
Set Destination = “sample extension”
When i calling from PSTN to asterisk (freepbx) asterrisk send messages - cli:
[2018-11-17 16:27:28] WARNING[5503][C-0000000d]: pbx.c:4416 __ast_pbx_run: Channel 'Dongle/dongle0-0100000008' sent to invalid extension but no invalid handler: context,exten,priority=default,720744425,1
-- Executing [h@default:1] Macro("Dongle/dongle0-0100000008", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("Dongle/dongle0-0100000008", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("Dongle/dongle0-0100000008", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] NoOp("Dongle/dongle0-0100000008", " monior file= ") in new stack
-- Executing [s@macro-hangupcall:5] AGI("Dongle/dongle0-0100000008", "attendedtransfer-rec-restart.php,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
-- <Dongle/dongle0-0100000008>AGI Script attendedtransfer-rec-restart.php completed, returning 0
-- Executing [s@macro-hangupcall:6] Hangup("Dongle/dongle0-0100000008", "") in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'Dongle/dongle0-0100000008' in macro 'hangupcall'
== Spawn extension (default, h, 1) exited non-zero on 'Dongle/dongle0-0100000008'
[dongle0] Got SMS from a2mobile: 'Uzytkownik numeru +48792093579 probowal sie z Toba polaczyc 17/11/2018 o 17:27. Jesli nie chcesz otrzymywac takich powiadomien, uzyj kodu *200*90#'
[2018-11-17 16:27:31] WARNING[5505][C-0000000e]: pbx.c:4416 __ast_pbx_run: Channel 'Local/sms@default-00000005;1' sent to invalid extension but no invalid handler: context,exten,priority=default,sms,1
-- Executing [h@default:1] Macro("Local/sms@default-00000005;1", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("Local/sms@default-00000005;1", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("Local/sms@default-00000005;1", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] NoOp("Local/sms@default-00000005;1", " monior file= ") in new stack
-- Executing [s@macro-hangupcall:5] AGI("Local/sms@default-00000005;1", "attendedtransfer-rec-restart.php,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
-- <Local/sms@default-00000005;1>AGI Script attendedtransfer-rec-restart.php completed, returning 0
-- Executing [s@macro-hangupcall:6] Hangup("Local/sms@default-00000005;1", "") in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'Local/sms@default-00000005;1' in macro 'hangupcall'
== Spawn extension (default, h, 1) exited non-zero on 'Local/sms@default-00000005;1'
dongle.conf
[general]
interval=15 ; 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=default ; 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=720744425 ; 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; no default value
data=/dev/ttyUSB2 ; 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
context=default
exten=720744425
imei=353871027734794
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
Regards - Karol
PS
Outbound calls work properly.