Helius Fax

Javier mira estos emails cortos..sobre Sangoma, Siemens, faxesHay muchas variables a ver, ademas de los problemas reales con los aparatos de fax (falta de papel, roturas, etc) y las limitantes actuales entre trixbox y la Siemens. Por eso creo, que seria muy importante tener un ambiente de pruebas..!!Y no pasar las penurias en el cliente como hoy...ademas de la falta de posibilidades por el tipo de conexion. ------------------------------Siememes HiPath 3550------------------------------------------------------No son los mismos valores que para Uruguay ojo!!Muy parecido a lo egregamos hoy al DryTek..!Estava tendo alguns problemas com uma Siememes HiPath 3550. Coloquei no indications.conf [br]description = Brazilringcadence = 1000,4000dial = 425busy = 425/250,0/250ring = 425/1000,0/4000congestion = 425/250,0/250,425/750,0/250callwaiting = 425/50,0/1000 ; Dialrecall not used in Brazil standard (using UK standard) dialrecall = 350+440; Record tone is not used in Brazil, use busy tonerecord = 425/250,0/250; Info not used in Brazil standard (using UK standard) info = 950/330,1400/330,1800/330stutter = 350+440 zaptel.conf defaultzone=brloadzone=br zapata.conflanguage=br sip.conf [general] country=br E resolvi o problema. ---------------------------Fax through Sangoma------------------------------------------------------- in our company we are trying to do this: Fax <--> Traditional PBX <--> Asterisk <--> PSTN In practice, we have put an Asterisk equipped with a Sangoma A102 (2 PRI ports) between our PBX ( Siemens HiCom) and the PSTN in order to have a VoIP network along the traditional telephony network. The problem is with the fax. We just want to send and receive faxes from/to our fax machine connected to the Siemens (without needing any interaction with our VoIP network, the faxes are sent to/received from PSTN). Unfortunately we are experiencing a lot of problems: the faxes not always work and when they work, it's likely to have incomplete pages. I know that faxing with VoIP is very troublesome, but maybe someone else is using a similar configuration and he found a good configuration or maybe has some hints to improve the results. We are using Asterisk 1.2.13.-------------------------Sangoma will detect the tones and disable echo cancel---------------------Incoming faxes, the Sangoma will detect the tones and disable echo cancel. To send outbound, you will have to add another trunk group, of one or more channels and disable echo cancellation and use that to dial out. Example (/etc/asterisk/zapata.conf)blah blahechocancel=yesblah blah group = 1channel =>1-20 blah blahecho cancel=nogroup = 2channel=22-23 So you would use for faxing specifically Zap/g2/ and it will use channel 22 or 23 but with echo turned off. Use Zap/g1 and it will use the first group of channels with echo cancel on (or whatever other parameters come before the group command)----------------------------------- ULAW ou ALAW ------------------------------------------------------Renato, FAX sobre VoIP é um negócio chato, nunca funciona 100%.Pense na possibilidade de colocar uma Placa FXS direto no seu Asterisk.DIGIVOICE -> ASTERISK -> FXS -> Aparelho de FAX.Assim pode ligar o FAX direto no Asterisk e não precisa converter sua chamada para VoIP e jogar na rede para o ATA, pois a latência,perda de pacotes e outros motivos influenciam diretamente no envio de FAX.Considerações,FAX e Multifuncionais modernas enviam FAX na velocidade de 33.6Kbps, enviando através de ATA / VoIP não vai funcionar.veja se consegue abaixar a velocidade no seu aparelho.Ative o recurso de correção de erro do seu fax (Se tiver)Lembre-se de sempre usar ULAW ou ALAW para envio de fax, se for outro codec com maior compressão não funciona.Certifique-se que está fixando o codec no sip.conf.Experimente desligar o cancelamento de echo do seu ATA, normalmente ele faz isso automaticamente ao detectar o sinal de fax, masé necessário forçar-----------------------------------T.38 support in Asterisk------------------------------------------The T.38 support in Asterisk 1.4 is only pass through. Asterisk is notan endpoint or gateway of any sort. It only works passthrough inSIP to SIP calls without chan_local or chan_agent involved. Both endpoints of the call will need T.38 support and have T.38enabled in sip.conf.Asterisk 1.4 as I see it is still a bit unstable and not ready forproduction use. Be careful out there.-------------------------------------------------------------------------------------------------------------------Podria seguir pegando Emails de Asterisk, trixbox y Siemens...pero no es el caso...Solo queria enviarte los que considere importantes..!! --------------------------------------------------------------------------------------------------------------------- "Nós somos o que fazemos no dia-a-dia. A excelência, portanto não é um ato, mas sim um hábito" - Aristóteles Grupo de Usuarios y Desarrolladores Debian del Uruguay Debian Uruguay (Guddú)http://debianuruguay.org/Debian (デビアン) は、あなたのコンピュータのためのフリーなオペレーティングシステム (OS) です しかし、OS の基本ツールのほとんどが GNU プロジェクトに由来しているため、GNU/Linux と名付けています。---