/ip firewall nat add action=dst-nat chain=dstnat dst-port=5060,10000-20000 in-interface=wan protocol=udp to-addresses=192.168..10 где 192.168..10 адрес вашего Elastix сервера wan порт на маршрутизаторе к которому подключен интернет.
(A->B). Reverse works. Instead, when in the same network (calls done from different softphones on my PC), all works more than OK. I repeat, A is external, B is an internal account of my asterisk configuration. Asterisk server is on a dedicated server on the net, with it's own IP, not behind NAT. Calls internal to internal work as they should. Installing from ISO: Debian for Elastix 5 Introduction. If you are not so familiar with Linux or Debian, then you can choose to install from ISO. In this case, Elasti 5 will install Debian 8 Jessie for you with the correct options, and subsequently install Elastix 5 as well. Hi Guys, Basically we have forwarded SIP Ports 5060 and 5061, RTP Ports 10000 - 10500. We're running elastix 1.6.0. Tried turning nat on and off but we cannot get any audio from a phone connecting from outside the network. Disable unneeded Asterisk modules. By default, Asterisk listens on many TCP and UDP ports as can be shown by netstat -anput | grep asterisk. If you only use SIP but not IAX2, and have no VoIP hardware cards, you can disable some Asterisk modules and close those ports. This could increase security in case your firewall goes down.
Configuration file for Asterisk SIP channels, for both inbound and outbound calls.. Starting with Asterisk v1.2.0: The global option "port" in 1.0.X that is used to set which port to bind to has been changed to "bindport" to be more consistent with the other channel drivers and to avoid confusion with the "port" option for users/peers.
NAT (Network Address Translation) is a technology most commonly used by firewalls and routers to allow multiple devices on a LAN with 'private' IP addresses to share a single public IP address. What is "Firewall and NAT traversal"? Select Asterisk-Cli. Type the following command: sip show registry; Click Execute button. Verify the state is Registered. Any other state indicates communications problem (firewall / NAT issue) between your Elastix server and GoTrunk network or incorrect Register string in your trunk configuration. Next follow "Routing configuration
OpenSIPS - Configuration and Integration with Asterisk (NAT Traversing) Mon, 11/17/2014 - 18:04 by falakniazi; OpenSIPS is an open source software that suits well for medium and large businesses to handle all the above mentioned scenarios. It supports common database access like mysql and postgres but still very well supports the carrier grade
Firewall & Router Configuration Overview - Brief overview of firewalls and ports with Elastix 5.0. Using the Firewall Checker - How to use the Firewall Checker utility embedded in Elastix 5.0. Routers, NAT and VoIP - Guide on the inner workings of NAT, PAT and why they are necessary. El NAT es la principal causa de problemas a la hora de montar nuestro servidor Asterisk. Desafortunadamente para nosotros, debido a la falta de IPs públicas de IPv4, lo normal en nuestros hogares es que estemos detrás de un NAT. Por lo tanto, si queremos montar nuestro Asterisk dentro de casa, tendremos que pelearnos con él. Instalación sencilla de Elastix 2.5 en una máquina virtual de Virtual Box. Procedimiento desde cero hasta la visualización web del sistema. nat=yes is working for asterisk version 10 or older. From asterisk 11 , nat=yes is depricated. They said nat=yes and nat=force_rport,comedia are same. But i think both are different. If we change to nat=force_rport,comedia the behavior seems to be fine, except for outside users behind NAT. Our server is also behind NAT. Elastix. Centralita telefónica en red. PBX. VoIP. - Juan Antonio Villalpando - Volver al índice del tutorial _____ - Utilizar los teléfonos desde el exterior de la red local. NAT . Pretendemos hablar desde un teléfono móvil en una ubicación remota con un Sotfphone ubicado en nuestra red local. Ambos configurados bajo el servidor Asterisk. Great article! I did have a problem getting it to work with my VOSP and Asterisk 1.4 (Asterisk and SIP clients behind a NAT router), though: In sip.conf, I had to have two sections (Outgoing and Incoming), and the Outgoing section had to be located before Incoming or I would get a BUSY signal when calling the VOSP number from a cellphone: