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Sds, Paulo Henrique Cardoso Administrador de Redes - T.I. NHS Sistemas Eletr?nicos Ltda Av. Juscelino Kubitschek de Oliveira, 5270 Cidade Industrial, Curitiba - PR Fone/Fax: (41) 2141-9246/9247 www.nhs.com.br IMPORTANTE: Esta mensagem, incluindo quaisquer anexos, ? endere?ada exclusivamente ao seu destinat?rio e poder? conter informa??es confidenciais. A revis?o, distribui??o, divulga??o e o uso n?o autorizado de tais informa??es ? proibido e estar? sujeita a penalidade cab?vel. Caso voc? n?o seja o destinat?rio, por favor informe o remetente respon...

Dear all , Is there any subset regression (subset selection regression) package in R other than "leaps"? Thanks and regards Alex [[alternative HTML version deleted]]

...t's the easiest way to do that via my TE110P on asterisk box. I know that is possible data transmission with this Digium Card, I'm wondering how... Any tip any tutorial? Probably someone around the world as already done this before. Best regards, Marco Mouta -- Esta mensagem (incluindo quaisquer anexos) pode conter informa??o confidencial para uso exclusivo do destinat?rio. Se n?o for o destinat?rio pretendido, n?o dever? usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is inte...

...ems to work fine in my Virtual Machine lab. The only issue i found was one module that is loaded with my modules.confthat I needed to copy from the backup /usr/lib/asterisk/modules and give the right permissions. Am I missing something? best regards, Marco Mouta -- Esta mensagem (incluindo quaisquer anexos) pode conter informa??o confidencial para uso exclusivo do destinat?rio. Se n?o for o destinat?rio pretendido, n?o dever? usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is inte...

Hi List; Is Digium the best telephony cards to be used with Asterisk? The prices are some how high, any suggestion? Regards Bilal ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

...7B350 SYSDOMAIN, Manutenção de Sistemas Informáticos, Lda. R. Luís de Camões, 39 CV Esq 1495-083 Algés Telf.: 214 194 354 Tlm: 917 215 972 Fax.: 211 454 920 Email: <mailto:clientes@sysdomain.pt> clientes@sysdomain.pt AVISO DE CONFIDENCIALIDADE: Este e-mail e quaisquer ficheiros informáticos com ele transmitidos são confidenciais e destinados ao conhecimento e uso exclusivo do respectivo destinatário, não podendo o conteúdo dos mesmos ser alterado. Caso tenha recebido este e-mail indevidamente, queira informar de imediato o remetente e proceder à destruição da me...

...nsidering 30 days per month 99,999% -> Max time for Outage during one month is 0,432 minutes If any of you around the world is aware of this values for VoIP SLAs I would be thankful to exchange and discuss this info. Thanks in advance. Best regards, Marco Mouta -- Esta mensagem (incluindo quaisquer anexos) pode conter informa??o confidencial para uso exclusivo do destinat?rio. Se n?o for o destinat?rio pretendido, n?o dever? usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is inte...

Dear all, I've created a digium certified asterisk professional - dCAP linkedin group for anyone, dCAP, interested: http://www.linkedin.com/e/gis/60298/39AE1350DBF3 Best regards, Marco Mouta dCAP November 2006 -- Esta mensagem (incluindo quaisquer anexos) pode conter informa??o confidencial para uso exclusivo do destinat?rio. Se n?o for o destinat?rio pretendido, n?o dever? usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is inte...

Hi, I have our software with SIP running on it.I configured asterisk server as proxy. How do I implement the call screening features(incoming and outgoing) using asterisk server.Please suggest me how to proceed on this. Thanks & Regards, Jahnavi. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps

Hello I want to use Asterisk to implement a SIP Domain allowing my clients to do URI dialing and receive calls from the Internet through URIs and ENUM. My question is, should I put my Asterisk outside the firewall (in the DMZ) to allow connections to the Internet? Or should I have it inside my local network and put a SIP Proxy (like Openser) in the DMZ to implement the SIP domain? Thanks

Hi, I am trying to implement call forwarding on the event that my ATA was not reachable to Asterisk, whether due to registration timeout, network interruptions between the ATA and Asterisk, or simply because the network on which the ATA resides in unreachable for any reason. I there a way of implementing such a feature in Asterisk? I have implemented CF unconditional, and CF on busy, CF on

We're looking at a large wifi phone deployment, and we're looking for wifi phones that: 1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally TFTP of own MAC address) 3. Industrial quality (no cheap plastic stuff). 4. Well documented (and none of the "only telco's get documentation" crap) Does anyone have a suggestion? Thanks, MD -------------- next

...asterisk 1.2.13) are causing all the troubles, making this multiple instances try to access same asterisk channel (leading us to Avoiding deadlock messages) ? I mean applying the patch might solve the problems instead off all system upgrade? Best regards, Marco Mouta -- Esta mensagem (incluindo quaisquer anexos) pode conter informa??o confidencial para uso exclusivo do destinat?rio. Se n?o for o destinat?rio pretendido, n?o dever? usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is inte...

...uot;valor da minha variavel) >same => n,Dial(SIP/10.68.2.43/${EXTEN},30,tT) >same => n,HangUp >Servidor 2: >exten => _X.,1,Answer() >exten => _X.,n,NoOp(${SIP_HEADER(Custom-variable)}) >exten => _X.,n,goto(ura,s,1) >exten => _X.,n,HangUp > >Voc? enviar quaisquer valores que possam ser definidos numa vari?vel. > >Neste sites voc? encontra maiores informa??es: >http://www.voip-info.org/wiki/view/Asterisk+cmd+SipAddHeader >https://wiki.asterisk.org/wiki/display/AST/Home > >O Jabber trabalha com o protocolo XMPP, de mensagens instant?neas.

...meira vez em Portugal no Outono de 2013 e terminará na Primavera de 2016. Esta formação é especialmente destinada a todos aqueles que trabalham com bebés (terapeutas de bebés, profissionais de terapia sacro-craniana, psicólogos infantis, pediatras, parteiras e especialistas em cuidados aos bebé) e quaisquer psicoterapeutas que têm experiência em trabalhar com adultos, particularmente em técnicas catárticas, e que gostariam de incluir o trabalho com bebés. Contudo, se a sua experiência é diferente e está interessado, por favor contate-nos para uma entrevista. 6 e 7 de Abril de 2013 Nos dias 6 e 7 de A...

We're going to be setting up Asterisk at our data center, as well as our call center locations via an optical fiber point to point connection. Is it best to have the servers communicate to eachother via SIP using SER, or just use the Asterisk functions? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi @ all, i set a server to a costumer of mine with a TE207P for use with 2 E1 Lines. I set them together into one group in zaptel/zapata.conf The point is now, the customer has a free-volumina of 60k minutes per month, per line. How can i make a kind of load balancing, that both lines will be trafficed the same way ? I read something about DIAL(Zap/r1/.) for using round robin, and

Hello. I am going through the documentation and trying to find if asterisk can help me in my case. It is quite difficult to find answer because I do not know the exact question. I have two location. Each in different country. Both locations have Siemens HiPath - different type and software. I can not use card that would allow me to connect those PBXs using SIP. But I have some free ISDN and

Good evening! I was wondering one thing, I'm using freepbx to configure my asterisk server and I have a problem with some inbound calls. When I receive a call to an INVITE sip:01xxxxxx at myip.com I an set an inbound route! It matches a DID number. How can I route an INVITE sip:s at myip.com? The number only appear in the To: Section. Thanks! Example: With this one, I cannot route it