tenta comentando: ;audio=/dev/ttyUSB2 ; tty port for audio connection; no default value ;data=/dev/ttyUSB3 ; tty port for AT commands; no default value e coloca somente o imei
2013/4/15 Otavio Asterisk <otavioaster...@gmail.com> > 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 < > ever...@visaotecnologia.com> escreveu: > > como esta o aquivo dongle.conf? >> >> >> Em 15 de abril de 2013 15:36, Otavio Asterisk >> <otavioaster...@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-unsubscr...@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-unsubscr...@listas.asteriskbrasil.org >> > > > > -- > 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-unsubscr...@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-unsubscr...@listas.asteriskbrasil.org