224 lines
7.0 KiB
Markdown
224 lines
7.0 KiB
Markdown
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<a href="https://travis-ci.org/wdoekes/asterisk-chan-dongle">
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<img alt="Travis Build Status"
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src="https://api.travis-ci.org/wdoekes/asterisk-chan-dongle.svg"/>
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</a>
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chan\_dongle channel driver for Huawei UMTS cards
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=================================================
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WARNING:
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This channel driver is in alpha stage.
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I am not responsible if this channel driver will eat your money on
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your SIM card or do any unpredicted things.
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Please use a recent Linux kernel, 2.6.33+ recommended.
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If you use FreeBSD, 8.0+ recommended.
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This channel driver should work with the folowing UMTS cards:
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* Huawei K3715
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* Huawei E169 / K3520
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* Huawei E155X
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* Huawei E175X
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* Huawei E261
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* Huawei K3765
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Check complete list in:
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http://wiki.e1550.mobi/doku.php?id=requirements#list_of_supported_models
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Before using the channel driver make sure to:
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* Disable PIN code on your SIM card
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Supported features:
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* Place voice calls and terminate voice calls
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* Send SMS and receive SMS
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* Send and receive USSD commands / messages
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Some useful AT commands:
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AT+CCWA=0,0,1 #disable call-waiting
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AT+CFUN=1,1 #reset dongle
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AT^CARDLOCK="<code>" #unlock code
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AT^SYSCFG=13,0,3FFFFFFF,0,3 #modem 2G only, automatic search any band, no roaming
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AT^U2DIAG=0 #enable modem function
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Building:
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----------
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$ ./bootstrap
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$ ./configure --with-astversion=13.7
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$ make
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If you run a different version of Asterisk, you'll need to update the
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`13.7` as appropriate, obviously.
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If you did not `make install` Asterisk in the usual location and configure
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cannot find the asterisk header files in `/usr/include/asterisk`, you may
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optionally pass `--with-asterisk=PATH/TO/INCLUDE`.
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Here is an example for the dialplan:
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------------------------------------
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**WARNING**: *This example uses the raw SMS message passed to System() directly.
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No sane person would do that with untrusted data without escaping/removing the
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single quotes.*
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[dongle-incoming]
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exten => sms,1,Verbose(Incoming SMS from ${CALLERID(num)} ${BASE64_DECODE(${SMS_BASE64})})
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exten => sms,n,System(echo '${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} - ${DONGLENAME} - ${CALLERID(num)}: ${BASE64_DECODE(${SMS_BASE64})}' >> /var/log/asterisk/sms.txt)
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exten => sms,n,Hangup()
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exten => ussd,1,Verbose(Incoming USSD: ${BASE64_DECODE(${USSD_BASE64})})
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exten => ussd,n,System(echo '${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} - ${DONGLENAME}: ${BASE64_DECODE(${USSD_BASE64})}' >> /var/log/asterisk/ussd.txt)
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exten => ussd,n,Hangup()
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exten => s,1,Dial(SIP/2001@othersipserver)
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exten => s,n,Hangup()
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[othersipserver-incoming]
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exten => _X.,1,Dial(Dongle/r1/${EXTEN})
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exten => _X.,n,Hangup
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exten => *#123#,1,DongleSendUSSD(dongle0,${EXTEN})
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exten => *#123#,n,Answer()
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exten => *#123#,n,Wait(2)
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exten => *#123#,n,Playback(vm-goodbye)
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exten => *#123#,n,Hangup()
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exten => _#X.,1,DongleSendSMS(dongle0,${EXTEN:1},"Please call me",1440,yes,"magicID")
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exten => _#X.,n,Answer()
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exten => _#X.,n,Wait(2)
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exten => _#X.,n,Playback(vm-goodbye)
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exten => _#X.,n,Hangup()
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You can also use this:
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----------------------
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Call using a specific group:
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exten => _X.,1,Dial(Dongle/g1/${EXTEN})
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Call using a specific group in round robin:
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exten => _X.,1,Dial(Dongle/r1/${EXTEN})
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Call using a specific dongle:
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exten => _X.,1,Dial(Dongle/dongle0/${EXTEN})
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Call using a specific provider name:
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exten => _X.,1,Dial(Dongle/p:PROVIDER NAME/${EXTEN})
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Call using a specific IMEI:
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exten => _X.,1,Dial(Dongle/i:123456789012345/${EXTEN})
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Call using a specific IMSI prefix:
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exten => _X.,1,Dial(Dongle/s:25099203948/${EXTEN})
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How to store your own number:
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dongle cmd dongle0 AT+CPBS=\"ON\"
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dongle cmd dongle0 AT+CPBW=1,\"+123456789\",145
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Other CLI commands:
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-------------------
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dongle reset <device>
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dongle restart gracefully <device>
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dongle restart now <device>
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dongle restart when convenient <device>
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dongle show device <device>
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dongle show devices
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dongle show version
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dongle sms <device> number message
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dongle ussd <device> ussd
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dongle stop gracefully <device>
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dongle stop now <device>
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dongle stop when convenient <device>
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dongle start <device>
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dongle restart gracefully <device>
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dongle restart now <device>
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dongle restart when convenient <device>
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dongle remove gracefully <device>
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dongle remove now <device>
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dongle remove when convenient <device>
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dongle reload gracefully
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dongle reload now
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dongle reload when convenient
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For reading installation notes please look to INSTALL file.
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Gain control and Jitter buffer
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--------------------------------
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<img src="https://cloud.githubusercontent.com/assets/6702424/26686554/9253bc18-46ed-11e7-9bce-cad8e2396435.png"
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width="800px" height="" />
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In order to perform good quality calls you will need to take care of:
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* **Automatic gain control**:
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chan_dongle does not control the gain of the audio stream it receive.
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This result of Alice hearing Bob's voice loud and noisy.
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It is possible to manually manage the gain in *dongle.conf* but
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the better option is by far to apply automatic gain control with
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the dialplan function AGC.
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* **Jitter buffer**:
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Since asterisk 12 it is no longer possible to enable Jitter buffer
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in dongle.conf it has to be applied in the dialplan.
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The lack of Jitter buffer result in severe loss in the transport
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of the voice from Bob to Alice.
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#### Dialplan example
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To set JITTERBUFFER and AGC in the dialplan on the appropriate channel
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regardless of who is initiating the call we will have to use
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the "b" option of Dial:
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b( context^exten^priority )
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Before initiating an outgoing call, Gosub to the specified
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location using the newly created channel.
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The Gosub will be executed for each destination channel."
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[from-dongle]
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; This will be executed by an indbound Dongle channel ( call initiated on the dongle side )
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exten => _[+0-9].,1,Dial(SIP/bob,b(from-dongle^outbound^1)) ;
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; This will be executed by an outbound SIP channel ( channel generated by dial )
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exten => outbound,1,Set(JITTERBUFFER(adaptive)=default)
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same => n,Set(AGC(rx)=4000)
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same => n,Return()
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[from-sip]
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; This will be executed by an inbound SIP channel ( call initiated on the SIP side )
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exten => _[+0-9].,1,Set(JITTERBUFFER(adaptive)=2000,1600,120)
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same => n,Set(AGC(rx)=4000)
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same => n,Dial(Dongle/i:${IMEI_OF_MY_DONGLE}/${NUMBER_OF_BOB})
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Note: To use automatic gain control dialplan function (AGC) you will need
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to compile Asterisk with func_speex ( see in menuselect ).
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On raspberry Pi you will need to compile and install speex and speexdsp yourself,
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the version of speex provided by the depos does not support AGC.
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(because compiled with fixed point instead of floating point)
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see: [HOWTO](https://gist.github.com/garronej/01f0dac45efe9161969a83890c019efa)
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For additional information about Huawei dongle usage look to
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chan\_dongle Wiki at http://wiki.e1550.mobi and chan\_dongle project home at
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https://github.com/wdoekes/asterisk-chan-dongle/
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