Distribution: linuxdebian. Its a Via C3 running at 800MHz with 512Mb RAM and a standard EIDE 80Gb disk. If I create a few extensions for my friends on my asteriskNow server , do I need to open a port ( Port Forwarding ) on my router so they can register their softphones on my asteriskNow server? 10-05-2006, 10:48 PM #2: farslayer. Posts: 7,247 Blog Entries: 5. Asterisk is the #1 open source communications toolkit. WWW: It is, in a sense, middleware between Internet and telephony channels on the bottom, and Internet and telephony applications at the top. To change the SIP port, open /etc/asterisk/sip.conf in your favorite text editor, look for the entry bindport and change the value of it to your new port number.
Opening ports on your router has security implications! Now, may I know the ports that I need to port forward to the DMZ box? What ports do I need to open for Asterisk VOIP? We have a similar situation, only we have NAT and we are using port forwarding on the Asterisk box.

Location: Northeast Ohio.

Please proceed with caution. This article explains what ports need to be open for remote phone and/or carrier connectivity, as well as the IP's of our SIP Trunking service to white-list and recommendations for SSH. It is instructed to establish a new connection to the resolved IP address and port. It’s NOT a firewall issue; no firewall is enabled on the Asterisk box. If no connection exists the first transport matching the transport type and address family as configured in pjsip.conf is chosen. Asterisk is an Open Source PBX and telephony toolkit. All handsets are running g711a (alaw which is standard for Australia) to the ITSP.

the PBX has an IP such as then you will need to perform additional configuration to allow Asterisk to route the SIP and RTP correctly. It is, in a sense, middleware between Internet and telephony channels on the bottom, and Internet and telephony applications at the top. The … Registered: Oct 2005. After which, do some port forwarding. An already open connection to the resolved IP address and port is searched for. If your Asterisk PBX is behind a NAT firewall, i.e. I would like to position the asterisk box behind the firewall. Thanks.
Remote Phones. Problem Description: Most conventional voip protocols (SIP, h323, …) are not programmed with NAT in mind, on itself they only carry call signaling (call setup, teardown,… and use RTP to carry the audio samples. The signaling usually uses fixed and standardized ports, but the RTP uses random ports to exchange both call legs (incoming and outgoing audio). I recon I could double it, judging by the overall CPU usage. The NAT configuration can be found in the file /etc/asterisk/sip.conf, the relevant section that needs to be edited is reproduced below: It is a good idea to change the default SIP port as most of the SIP vulnerable attacks occurs on it’s default port 5060. If the connection exists it is reused for the request. Hello, I am new to voip and asterisk and I not sure if my statement below is required or not. It manages to keep up with 6-8 calls and no complaints. SIP with NAT or Firewalls. I can run tcpdump on the Asterisk box, which confirms that I can get the extensions to send SIP commands to Asterisk, but there is no response whatsoever from Asterisk because port 5060 is not open. LQ Guru . 1.1. Proper Ports to Open for SIP and RTP. Asterisk powers IP PBX systems, VoIP gateways, conference servers, and is used by SMBs, enterprises, call centers, carriers and governments worldwide.