For FreePBX versions 2.5 or newer, we have a simple module included in FreePBX called "SIPStation," which makes setting up your trunks a breeze. You can view this wiki on how to use the SIPStation module.
If you cannot get the SIPStation module to work or would prefer to manually create your trunks, please see our wiki "Setting up Standard SIPStation Manually in FreePBX."
If you are using another Asterisk-based PBX, then you should be able to use the configuration instructions available in our wiki "Setting up Standard SIPStation Manually in FreePBX." Simply make substitutions to the instructions as needed for your specific Asterisk system. We do not currenlty have configuration instructions for other systems, but there are plenty of users connected to our service with other systems, so you should usually be able to get the service working. We require a SIP registration to receive inbound calls. We require a challenge/response authentication for outbound calls. If your system is capable of this configuration (as most SIP capable systems are), then you should be able to configure our service.
Please see our step-by-step wiki "Setting up Standard SIPStation with SwitchVox" to learn how to use your SwitchVox system with SIPStation.
We recommend forwarding ports UDP/5060 and UDP/10000-20000 for standard FreePBX/Asterisk-based installs. If using newer versions of FreePBX, port 5160 is the default port for ChanSIP so that may be the port you need to forward. Check Asterisk SIP Settings for the bind port of ChanSIP. It may be possible to get your service working without port forwarding, but optimal service will be obtained with the above mentioned ports. You can lock down port UDP/5060 or UDP/5160 depending on bind port of ChanSIP to the trunk1.freepbx.com and trunk2.freepbx.com FQDNs for additional security, but please note, we do from time to time change the IP addresses associated with these FQDNs. Therefore it is best to use the FQDN and not an IP Address. You cannot lock down UDP/10000-20000 to any specific IP address, since the media of a phone call can come from hundreds of different IP addresses.
You can lock down port 5060/UDP to trunk1.freepbx.com and trunk2.freepbx.com. You cannot lock down the media ports because the media servers vary and change. Most security issues that are reported are usually related to manufacturer vulnerabilities in their SIP stack, port 5060. By locking down this signaling port, you should be able to address almost all potential issues.
We support g711 (ulaw) and g729.