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. Inside EPM you define per template if the phones use TFTP, FTP, HTTP or HTTPS provisioning. Recommend using HTTPS Phone Provisioning option of instead for remote users. Not recommended to open this up to untrusted networks as the traffic is not encrypted. Used by UCP with HTTPS for Conf Rooms and Chatting and other products in UCP 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 Safe to open this up to untrusted networks as the traffic is encrypted with SSL and requires username and password authentication.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 Port 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. If the issue still persists after opening both ports locally on the server, contact your service provider to find out where the connection is blocked.Can change this port inside the PBX Admin GUI SIP Settings module. Make sure that firewall settings on the source server allow connections via TCP ports 22/21. Telnet: connect to address 203.0.113.2: Connection timed outĬonnections to ports 22/21 are blocked by a firewall on the source server. When connecting to ports 22 or 21 (Plesk Website Importing tool) on the source server using the telnet utility, the connection fails with: PLESK_ERROR: Failed to connect to source domain by FTP: timed out PLESK_ERROR: Failed to connect to source domain by SSH: Unable to connect to ‘203.0.113.2’ by SSH: timed out. PLESK_ERROR: Failed to check SSH connection to the source server ‘source’ (203.0.113.2): Unable to connect to ‘203.0.113.2’ by SSH: Connection timed out.Įnsure that the server is up and there are no firewall rules that may block SSH connections to the server, then restart migration. Plesk Migrator or Plesk Website Importing tool fails to connect to a source server with one of the following error messages:
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