search for: embratel

...=201-238 fxsls=239-254 /etc/asterisk/chan_dahdi.conf: [trunkgroups] [channels] language=pt_BR group=1 switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown priindication=outofband signalling=pri_cpe echocancel=yes echocancelwhenbridged=no relaxdtmf=no switchtype=euroisdn context=e1-embratel group=1 signalling=pri_cpe channel =>1-15,17-31 switchtype=euroisdn context=e1-bp250 group=2 signalling=pri_net overlapdial=yes channel =>32-46,48-62 switchtype=euroisdn context=e1-embratel group=1 signalling=pri_cpe channel => 63-77,79-91 switchtype=euroisdn echocancel=yes context=e1-e...

...ando um ambiente de laborat?rio para entender melhor todo a quest?o de uso e configura??o da ferramenta. Bom o meu problema esta no seguinte:--------------------------------------------------- Tenho uma placa FXO com 4 canais anal?gicos (Troco), onde estou conectando minhas 4 linhas da anal?gica da Embratel. At? ai tudo bem, consigo fazer liga??o externa sem problemas. Meu problema come?a porque tenho configurado, pela Embratel, o servi?o de Busca Autom?tica, ou seja, liga-se no troco ele toca uma vez em cada uma das outras linhas. Nesse cen?rio o meu Asterisk esta encaminhando a chamada para a outra...

...e" from C code. The intention of the research is to create a DLL. Is it possible and where can I find an example? Sorry for bad english! ___________________________________________________________________________________ Acesse nosso portal www.click21.com.br Porque internet grátis, nem a Embratel pode fazer mais barato. Mas pode fazer melhor.

...lling system and I need to charge my customers for collect calls they receive. Currently I have to get the PSTN billing and locate all collect calls and then enter it manually. I want to detect these calls in asterisk so I can automatically charge my customers. For Brazilians in the list, I have a Embratel E1 - ISDN. Thanks Carlos Barros

Hi all, I am new to this list... Can I use Asterisk as a SIP Proxy and as a SIP to H.323 translator? I want to implement PC-to-Phone calls in the following topology (for signalling): SIP Softphone --> Asterisk --> Gatekeeper H.323 ---> Gateway H.323 ---> PSTN The RTP audio packets would go direct through Softphone to gateway. Does someone have a configuration file example of

...; to change in the config is the variant for your contry thats br. > > Keep us post in your progress so we can know that R2 is working in > other countrys! > > On 05/10/2004, at 6:45 AM, Miguel wrote: > >> Miguel, >> >> I'm in Brazil and I'll try it with Embratel (a telmex company), may >> you send >> me your configs ? Are there any special advice to implement it ? >> >> Kind regards, >> >> Miguel Antonio >> >> Date: Mon, 4 Oct 2004 15:18:22 -0500 >> From: Miguel Cavazos <miguel@cavazos.com.mx> &g...

Hi all I am beginning in asterisk and am making tests with an ata-186. For the time being the tests are going well, however have a doubt. I am thinking about using a canal e1 with plate digium. Assuming that the company of telecommunications supplies e1 with 30 canals and numeration to me 4000-0001 4000-0029. she is possible to configure asterisk in way that somebody of is dials 4000-0025, to

Our ISP experienced backbone problems yesterday. Our backbone provider's backbone provider (EMBRATEL, the newly-privatized telecom company of Brazil) went down for a few hours. That was a big problem itself, but what I'll tell you is a weird side effect: Samba stopped working on our local network!!! I checked DNS, resolv.conf, /etc/hosts, smb.conf, everything! While our Internet connection w...

Miguel, How many simultaneous calls (incoming and outgoing) you did ? Kind regards, Miguel Date: Sun, 3 Oct 2004 11:21:04 -0500 From: Miguel Cavazos <miguel@cavazos.com.mx> Subject: [Asterisk-Users] Working E1 MFC/R2 M?xico !!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A)----> Cisco AS5xxx ----> ASterisk---->PSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains

Hi All, as good? I am trying to make a call for the Unicall channels and after the exchange of signalling and sending of the digits asterisk locks up with the sending of the signalling "E" and the TELCO says that asterisk would have to send signalling "F", as to make for asterisk to send signalling "F"? The TELCO says that the signalling "E" is