SIP Trunking with sipXecs: Configuration

Configure SipXbridge

Navigate to Devices>SBC.
Select the sipXbridge-1 instance defined and configure it. Note the Incoming calls destination setting ( defaults to operator ). This is a convenience field. You can set this to a hunt group extension, conference extension or other extension that is not an alias for a real user. Leave this field blank and use your DID as a user alias to direct your call to a specific user alias. SIPX has powerful user alias based call routing capabilities and this is the preferred way to route calls to endpoints that have user accounts associated with them. You would want to do that, for example, if you have multiple DIDs from the same provider and you want each DID to be assigned to a different user.

The Public port in this page is the port that is exposed to the public network through your firewall setting. If your firewall restricts inbound traffic, you must open this port on your firewall to allow inbound signaling from the ITSP. The External port, is the port that is a port on the machine that sipxbridge runs on. It "faces" the firewall. It is associated with the public port on the firewall. Hence the firewall must be configured to send packets from the the public port to the external port. If you leave the public port blank, the external port is assumed to be the same as the public port (i.e. the mapping is assumed to be symmetric). If your firewall filtering rules allow inbound traffic from those destinations to which outbound traffic has previously been sent and if your ITSP provides "hosted NAT compensation", you do not need to reconfigure any firewall rules.

SipXbridge runs on port 5080 (not 5060). You can change port on which it receives signaling. However, if you change the sipXbridge port, be careful of causing port conflicts with other sipx components that are co-located on the same platform that bind to the same IP address. The port where sipxbridge expects to recieve signaling has nothing to do with where the ITSP expects to receive its signaling. The ITSP can continue to receive its signaling at port 5060. If your ITSP does IP address provisioning (i.e. ITSP registers your public address and signals that public address), they will probably default to signal sipXbridge on port 5060. If you do a straight through mapping on your firewall (i.e. external port maps to identical internal port) and open up port 5060, the signaling from the ITSP would bypass SipXbridge and go directly to the SIPX Proxy server and hence SipXbridge would not work. Please contact your ITSP and provision their system to signal port 5080 on your public address and open up port 5080 on your firewall (recommended) or use appropriate firewall rules to map external port 5060 to port 5080. If you chose to do the latter (not recommended - especially if you are also configuring remote workers), you would need to specify what port on the firewall you have mapped in the screen above. This note does not apply to ITSPs that function by Registration.

Typically ITSPs do not handle certain types of SIP requests such as REFER which is used in Call Transfer operations. To implement call transfer, SipXbridge does signaling translation, converting a REFER request to an INVITE request to the call transfer target. Consequently, a ringing tone will not be heard at the calling phone during call transfers when the call is routed through SipXbridge. Enable Music On Hold (MOH)on this page if you would like to hear music for blind transfers. If you do not do this, you will hear silence during the time a call is being transferred blind. You are recommended to turn MOH off for your phone when MOH is turned ON on sipXbridge as certain signaling race conditions may occur, resulting in garbled MOH.

Configure NAT Traversal

Navigate to System > Servers > NAT. This will take you to a page where you can configure your NAT traversal service settings. You can select to use STUN or enter your public address here. A relay service (known as SipXrelay) manages a range of ports which defaults to the range 30000 to 31000. This setting must be a contiguous range of free ports.

If your server is running behind a NAT you must also explicitly declare that. Go over to System > Internet Calling and select the NAT Traversal Link. Check the Server Behind NAT box. If you plan to configure remote workers you should also enable NAT traversal on this page.

Configure a Dial Plan

Navigate to System > Dial Plans. Using the pull down menu, define a new Dial plan.
In the Gateways section drop down list, select the action to add a new SIP Trunk Gateway.
Specify a name, the address of the ITSP (i.e. 204.11.192.31 for callcentric). You should see the previously defined SBC (i.e. sipXbridge-1) appear in the drop down list for the SBC Route.
After you are done adding the Gateway, you must select the "Enabled" check box. Click on Accept and OK to back out of this screen.

Configure an ITSP account that is managed by SipXbridge

Most ITSPs only need for you to specify a proxy domain, user name and password. User Name is mandatory for accounts that require Registration with the ITSP.

Many ITSPs allow web access to set up your account. The password on this screen is your SIP password and not your web account password.

Some ITSPs may require advanced settings. To enter these settings, you can navigate to the ITSP Account settings from the gateway screen. For example, the Asserted-Identity field may be specialized. Click on the Advanced link to change these settings.

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