similar to: SIP trought Firewall

Hello, I am configuring apache 2.2 to serve my rails app through mongrel at the root of my server. First, I redirect all http traffic to https with the following: ServerName emiliano NameVirtualHost *:80 <VirtualHost *:80> RedirectPermanent / https://emiliano #this server is only accessed internally </VirtualHost> Inside the <VirtualHost *:443> section, I have the following

Right now I've been working on setting up an Trixbox server on our internal network. Its behind the firewall, but I'd like to open up the firewall to it because we sometimes have developers working off site and I'd like them to be able to connect. Is this safe to do? I've got the "Allow Anonymous Inbound SIP Calls" box unchecked in freePBX. Is there anything else

---------- Forwarded message ---------- From: Emiliano Zapata <ezapataika@gmail.com> Date: Sun, May 20, 2012 at 12:09 PM Subject: To: R-help@r-project.org Hi, I have a 64 bits machine (Windows) with a total of 192GB of physical memory (RAM), and total of 8 CPU. I wanted to ask how can I make R make use of all the memory. I recently ran a script requiring approximately 92 GB of memory to

Hello, using Asterisk 1.4.28, I encountered a problem with SIP RTP port allocation. I found some entries in mailinglist and bugtracker regarding this issue, but only old ones. My rtp.conf has [general] rtpstart=30000 rtpend=30100 so 100 ports available. I know that up to 4 ports per channel can be used and so up to 25 channels are possible. But even earlier I often get the error about

After getting some ERRORS like this: [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports

Hi! I have a question about agents in asterisk. In first place, agent login to asterisk (from the telephone) The question is: Can an agent take a break (using a function *(some number)) from the phone? Thanks to all Regards -- Ing Francisco Roqu? 3Tech SRL Plaza Paso N?92, EP B Buenos Aires, Argentina. www.3tech.com.ar

Hi! I have a chan dongle and I want to use PlaySMS with Chan Dongle for send many SMS per day. Is possible to use this? Thanks

Hola me interesar?a realizar esa conexi?n, para poder manipular las tablas Access desde R. Alguien sabe el procedimiento y que paquetes debo bajar y si hay alguna otra especificaci?n t?cnica respecto a la configuraci?n del Windows en 64 bits? Saludos Enviado desde Correo<https://go.microsoft.com/fwlink/?LinkId=550986> para Windows [[alternative HTML version deleted]]

> Can anyone comment on using SMS in conjunction with VoIP service using > one of these three VoIP providers: voip.ms, vitelity.com, > flowroute.com? Are some SMS services more compatible with Asterisk > (i.e. SMS over SIP works perfectly or not)? Is it best to use a > different data channel for SMS messages (i.e. SMS via HTTP, SMS via > XMPP) instead of Asterisk's built

Hi list I'm kinda new to asterisk and I'm woriking for a company that sells Asterisk solutions and appliances. I installed TrixBox on a litle PC @ home and a x100p card which is recognized as a Zaptel card, I made some in/outbound routes and they seem to work but I have a problem with SIP softphones. I created 2 estensions 1000 and 1001 they're both in different cities, when I 1000

Hello dear mailing list readers, Recently I am experimenting a problem trying to retrieve messages over a POP3 protocol. Everything goes right until the POP3 client tries to retrieve a message that contains one or more long lines ? lines as long as 3000 characters or even more. The client (Outlook 2003 and 2007) stops on that message until timeout. I have opened a connection manually to see

Estimado Emiliano De Iorio ¿Que posibilidades hay de tomar access y pasarlo a otro base de datos, o sqlite para tener todo en un archivo? Si puede migrar los datos ganaría en una cantidad de herramientas disponibles en R para trabajar con los datos. Javier Rubén Marcuzzi > El 7 ene 2024, a las 16:59, emiliano di iorio <emilianodiiorio en hotmail.com> escribió: > > Hola me

Hi, I want to update two divs simultaniously in a page. Can anyone help me please? Emiliano

Hi, I need to set up a weird dovecot configuration: 1) outside a ssl tunnel I'd like to authenticate only with cram-md5 scheme 2) inside a ssl tunnel I'd like to authenticate only with plain auth The first is easily satisfied with auth_mechanisms = plain cram-md5 disable_plaintext_auth = yes but I don't know how to satisfy the second condition, if it's possible. Thanks for

El 29/03/16 a las 08:29, Steve Howes escribi?: > I don't think you can. Whatsapp is a closed system. > > Steve And they change your code every day and make it always obfuscated. https://github.com/tgalal/yowsup/issues/887 Best regards. Emiliano.

Hi, I'd like to subscribe folder with doveadm: doveadm mailbox subscribe -u rago public.Conferences This command doesn't work, while it works with an ordinary folder. However it's possible to subscribe to the folder with an imap connection: 1 login rago "mypasswd" 1 OK [CAPABILITY IMAP4rev1 LITERAL+ SASL-IR LOGIN-REFERRALS ID ENABLE IDLE SORTSORT=DISPLAY

On 2/23/19 4:19 PM, Joshua C. Colp wrote: > On Sat, Feb 23, 2019, at 11:04 AM, hw wrote: > > <snip> > >> >> directmedia is not explicitly enabled; I guess it's the default. >> >> Joshua basically says there is no way to control which ports are being >> used for SRTP because that it is "up the endpoint". Such endpoints, in >>

Hi all, I'm new to the list, so ... hello everybody :-) Let's go to the point... I'm implementing quota over imap support in my mail server, so since a number of days I'm tring to upgrade to the new 1.x version (debian sarge) ... After having a number of troubles using the debian backport packege I am now using the -rc28 package that I have built by myself... I installed

Hello. I'm trying to test an Asterisk server by using a VOIP provider for international calls but, I'm having problems trying to get my server communicate with theirs. I don't know if I'm having all these issues becuase I'm behind NAT or what. I have the following in my server's sip.conf: [provider] type=peer host=<theprovider's server> username=<username>

We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching