Sangoma Connect Signaling Sangoma Connect uses chan_PJSIP TCP signaling by defaultĬhange this port in the PBX Admin GUI → Settings → Asterisk SIP Settings → PJSIP TCP Bind Port Inside EPM you pick if the phone apps use HTTP or HTTPS. Port used for phone apps to communicate with the PBX using SSL encryption. Safe to open this up to untrusted networks as the traffic is encrypted. Port used for phone apps to communicate with the PBX as HTTP not HTTPS traffic. Recommend using Phone Apps HTTPS option of instead for remote users. Not recommended to open this up to untrusted networks as the traffic is not encrypted. Not recommended to open this up to untrusted networks as it has no ability encrypt traffic and is not NAT Friendly. Inside EPM you define per template if the phones use TFTP, FTP, HTTP or HTTPS provisioning. Used if your are having phones inside EPM use TFTP for provisioning. Recommend using HTTPS provisioning for remote phones instead. This can only be changed inside from inside Linux CLI and not recommended to be changed. Make sure if opening up outside access to enable username and password authentication for HTTP provisioning from the PBX Admin GUI System Admin > Provisioning Protocol. Safe to open this up to untrusted networks as the traffic is encrypted as long as your enable username and password authentication as outlined in the Notes section. When upgrading older systems, the port assignments to not change from their original settings. In the past, http provisioning defaulted to port 83. Recommend using HTTPS Phone Provisioning option of instead for remote users. Used by UCP with HTTPS for Conf Rooms and Chatting and other products in UCP Safe to open this up to untrusted networks as the traffic is encrypted with SSL and requires username and password authentication. Used by UCP with HTTP for Conf Rooms and Chatting and other products in UCPĬan change this port inside the PBX Admin GUI > Advanced Settings > UCP NodeJS Server > NodeJS HTTPS Bind Port Traffic is encrypted with SSL and requires username and password authentication, so verify using very complex username and password.Ĭan change this port inside the PBX Admin GUI > Advanced Settings > UCP NodeJS Server > NodeJS Bind Port Recommend using HTTPS versionĬan change this port inside the PBX Admin GUI > Advanced Settings > Asterisk Builtin mini-HTTP section > HTTPS Bind PortĬautiously open this up to untrusted networks if needed. Port used to access the GUI portion of UCP with SSL encryptionĬan change this port inside the PBX Admin GUI > Advanced Settings > Asterisk Builtin mini-HTTP section > HTTP Bind Port Safe to open this up to untrusted networks as the traffic is encrypted and requires username and password authentication. Port used to access the GUI portion of UCP Recommend using HTTPS version of PBX User Control Panel instead for remote users. PBX User Control Panel (UCP) HTTP (Non HTTPS)Ĭan change this port inside the PBX Admin GUI > System Admin Module > Port Management section. Not configurable in the GUI, on by editing custom conf file. Used for the actual voice portion of a SIP Call.Ĭan change this port inside the PBX Admin GUI IAX Settings module. Safe to open to the outside world and is required by most SIP Carriers as your RTP traffic can come from anywhere. Secure Port used for chan_SIP Signalling. Standard Port used for chan_SIP Signalling. Secure Port used for chan_PJSIP Signalling. Standard Port used for chan_PJSIP Signalling. Not recommended to open this up to untrusted networks. Can change this port inside the PBX Admin GUI SIP Settings module.
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