本帖最後由 角色 於 2013-11-3 23:17 編輯
In order to get the system work, you have to add/modify the following variables in /etc/asterisk/dongle.conf
Listing 1- [defaults]
- context = DID_3Gdongle0 (This context will be used in Asterisk-GUI)
- .
- .
- .
- exten = +852RRRRSSSS (SIM card telephone number)
- [dongle0]
- imei = XXXXXXXXX453218 (the IMEI code of the SIM card)
- imsi = YYYYYYYYY232505 (the IMSI code of the SIM card)
複製代碼 In /etc/asterisk/dongle.conf
Listing 2- ;!
- ;! Automatically generated configuration file
- ;! Filename: dongle.conf (/etc/asterisk/dongle.conf)
- ;! Generator: Manager
- ;! Creation Date: Sun Nov 3 19:15:14 2013
- ;!
- [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]
- context = DID_3Gdongle0
- group = 0
- rxgain = 0
- txgain = 0
- autodeletesms = yes
- resetdongle = yes
- u2diag = -1
- usecallingpres = yes
- callingpres = allowed_passed_screen
- disablesms = no
- language = en
- smsaspdu = yes
- mindtmfgap = 45
- mindtmfduration = 80
- mindtmfinterval = 200
- callwaiting = auto
- disable = no
- initstate = start
- exten = +852RRRRSSSS
- dtmf = relax
- ; 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
- ;imei=123456789012345
- ;imsi=123456789012345
- imei = XXXXXXXXX453218
- imsi = YYYYYYYYY232505
- ; 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
複製代碼 The the following /etc/asterisk/users.conf was created by Asterisk-GUI and modified by hand
Listing 3- [3Gdongle0]
- trunkname = 3Gdongle0
- context = DID_3Gdongle0
- hasexten = no
- hasiax = no
- hassip = yes
- registeriax = no
- registersip = yes
- trunkstyle = voip
- disallow = all
- allow = all
複製代碼 The listing 4 (/etc/asterisk/extensions.conf) was first initialised by Asterisk-GUI and modified by hand
Listing 4- [CallingRule_Dongle0]
- ;exten = _82X.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${EXTEN:2},,trunk_1,)
- exten = _82X.,1,Dial(Dongle/dongle0/${EXTEN:2})
- exten = _82X.,2,Hangup()
複製代碼 |