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  • Changing the SIP Domain Name for Sipxcom in Release 17.04
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Introduction

In Sipxcom, the fully qualified name for a voice server is defined at setup time by the host name (e.g. pbx) and SIP domain name (e.g. lvtest.com). After first installing the Sipxcom ISO, the server will restart. When logging into root after the reboot, the sipxecs-setup script is automatically started which prompts the user whether to change network settings, if this is the first Sipxcom server, and prompts for hostname and SIP domain. After a few minutes, sipxecs-setup completes and displays message to  log into the voice server via a web browser.

 

In older releases of Sipxcom, the voice server fully qualified domain name (e.g. pbx.lvtest.com) was changed by shutting down Sipxcom from the root account, and re-rerunning the sipxecs-setup script. In release 17.04, it appears sipxecs-setup fails to update the Mongo database name to the new updated fully qualified name for the voice server. This document describes how to complete the conversion of the fully qualified domain name in Sipxcom from pbx.lvtest.com to pbx.lvtest1.com.

Step 1 - Shut Down all Sipxcom Processes from Root

SSH into the voice server as root and issue the following commands:

  • service sipxecs stop
  • service sipxsupervisor stop
  • service mongod stop
  • service postgresql stop
  • crontab -r to stop any automated Sipxcom processes from starting
  • Do a ps -ef | grep sipx command and issue a kill -9 to any remaining Sipxcom processes
  • Do a service sipxcom status command to ascertain all major Sipxcom processes are stopped

Step 2 - Run sipxecs-setup to Change FQDN of Voice Server

Run the sipxecs-setup script to change the Sipxcom FQDN from pbx.lvtest.com to pbx.lvtest1.com:

Step 3 - Check Whether Sipregistrar is Running

Check the running Sipxcom processes via a service sipxecs status command - if the sipxsaa, sipxrls, sipstatus, and sipregistrar processes are stopped, then this is due to the Mongo hostid not being converted by sipxecs-setup from pbx.lvtest.com to pbx.lvtest1.com. Do a tail command on the /var/log/sipxpbx/sipregistrar.log file - there will be log messages for connect errors to the Mongo pbx.lvtest1.com database.

Step 4 - Convert FQDN in Mongo from pbx.lvtest.com to pbx.lvtest1.com


Go into Mongo and issue the following commands - upon completion, exit Mongo and restart the Sipxcom server:

  • Issue Mongo command
  • Issue rs.config() command - you will notice that host name is still pbx.lvtest.com
  • Issue the cfg = rs.config() command - which copies the database parameters into a variable

  • Issue the cfg.members[0].host = "pbx.lvtest1.com" command which changes the voice FQDN host name to pbx.lvtest1.com in Mongo

  • Issue the "rs.reconfig(cfg,{force:true})" command to apply the new pbx.lvtest1.com host name to Mongo

  • Issue the rs.config() command again to ascertain the new host name has been applied in Mongo

  • Issue the exit command from Mongo and then restart the Sipxcom server

Step 5 - Check Intranet Domain Field in the Internet Calling Menu

Go to the System→Internet Calling or System->Settings→Internet Calling menu, and if necessary, update the Intranet Domain field with the new lvtest1.com domain.

Step 6 - Push Sipxcom Server Processes and Validate All Processes are Running

Push the pbx.lvtest1.com server profile which copies all configuration data from the SQL server into Mongo. Check to ascertain all server processes are running.

Step 7 - Rebuild Phone Profiles and Restart Phones

The phones registered to the Sipxcom voice server currently use lvtest.com as the SIP domain - push all phone profiles which rebuilds the configuration files on the Sipxcom TFTP directory. The phones will need to be manually restarted to pick up the new configuration files and register to the voice server with SIP domain lvtest1.com.

Step 8 - Test Incoming, Outgoing Calls, Voicemails, Autoattendants, etc

Test a variety of internal and external calls, call forwards, voicemail, autoattendants to ascertain all calls are working properly. If bearer path fails to appear on some calls (e.g. voicemail announcements), and you are testing with a new voice server, then pay attention to your NAT traversal settings:

  • If this is a new system and Sipxbridge is used,for external calls, ascertain the NAT Traversal type is set to IP address, and Public IP address is set to the IP address assigned to the WAN router, assuming Sipxcom is behind a firewall.
  • If this is a new system and unmanaged gateways are used for external calls, then pay attention to the following settings, particularly if voicemail or autoattendant announcements disappear after 30-60 seconds (assume Sipxcom and gateway is behind a firewall):
    • NAT public IP address should be configured to be the Sipxcom private IP address.
    • The Enable NAT traversal and Server behind NAT settings should be disabled.


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