[AsteriskBrasil] Elastix + Dongle 3g
Otavio Asterisk
otavioasterisk em gmail.com
Segunda Abril 15 16:33:11 BRT 2013
praticamente default, alterei apenas o número da usb, e o meu imei e imsi.
[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=from-pstn ; 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=+1234567890 ; 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/ttyUSB2 ; tty port for audio connection; no
default value
data=/dev/ttyUSB3 ; 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=XXXXXXXXXXXXXXX
imsi=XXXXXXXXXXXXXXX
; 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
Em 15 de abril de 2013 15:41, Everton Carneiro
<everton em visaotecnologia.com>escreveu:
> como esta o aquivo dongle.conf?
>
>
> Em 15 de abril de 2013 15:36, Otavio Asterisk <otavioasterisk em gmail.com>escreveu:
>
>> Lista, boa tarde.
>> Consegui subir o modem, consigo realizar e receber ligações, no entanto,
>> qnd ligo através do modem, a pessoa q me atende não ouve nada, apenas um
>> chiado e alto e um pouco da minha voz metalizada. O mesmo ocorre qnd ligo
>> para o chip q está no modem: ouço a ura do meu pabx toda metalizada e com
>> muito chiado de fundo.
>> Alguém já passou por isso ou tem alguma dica?
>> Abraço a todos!
>>
>> --
>> Otávio
>>
>> _______________________________________________
>> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1;
>> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7;
>> Intercomunicadores para acesso remoto via rede IP. Conheça em
>> www.Khomp.com.
>> _______________________________________________
>> DIGIVOICE Fabricante de Placas de Voz e Channel Bank
>> 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
>> Centro Treinamento - Curso de PABX IP - Asterisk - Site
>> www.digivoice.com.br
>> _______________________________________________
>> ALIGERA – Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7.
>> Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.
>> Channel Bank – Appliance Asterisk - Acesse www.aligera.com.br.
>> _______________________________________________
>> Para remover seu email desta lista, basta enviar um email em branco para
>> asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>>
>
>
>
> --
> *Everton Carneiro .:*
> *Visão Tecnologia
> *
> *Fortaleza-CE 85-3044 8888 / 3044-8844
> *
> *Cel: Tim 85-9665 0888
> *
>
> Preserve o verde, antes de imprimir veja se realmente é necessário.
>
> _______________________________________________
> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1;
> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7;
> Intercomunicadores para acesso remoto via rede IP. Conheça em
> www.Khomp.com.
> _______________________________________________
> DIGIVOICE Fabricante de Placas de Voz e Channel Bank
> 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
> Centro Treinamento - Curso de PABX IP - Asterisk - Site
> www.digivoice.com.br
> _______________________________________________
> ALIGERA – Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7.
> Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.
> Channel Bank – Appliance Asterisk - Acesse www.aligera.com.br.
> _______________________________________________
> Para remover seu email desta lista, basta enviar um email em branco para
> asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>
--
Otávio
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