The rerouting feature can be turned off and on systemwide, and the SIP response codes upon which rerouting is provided are provisionable in the system. A rerouting timer also provides rerouting if no response is received from the SIP gateway, SIP server, or SIP-Q gateway after an INVITE has been sent.
SIP calls can be rerouted if:
A gateway cannot process a connection request. The calls can be of type offnet (to the PSTN (Public Switched Telephone Network) via a SIP gateway) or on-net (to another SIP or SIP-Q network such as a QSIG gateway or the Lotus Sametime UC (User Client) Application).
A WAN failure occurs. SIP and SIP-Q calls between subscribers are rerouted through the PSTN.
A SIP response code indicates a bandwidth restriction.
Requirements for rerouting calls
For a call to be rerouted, the calling subscriber must either be calling from a different survivable branch, or be directly registered with Lotus Sametime Unified Telephony. The called subscriber must:
Be registered from a survivable branch.
Reside in that survivable branch. This means that the called subscriber must be registered with its provisioned survivable IP EP (Endpoint). This EP is administered through Lotus Sametime Unified Telephony Assistant.
Have a valid public E.164 number.
When a call is rerouted through the PSTN, the resulting connection consists of the following:
The originating call leg from the calling SIP telephone to the outgoing PSTN gateway
The terminating call leg from the incoming PSTN gateway or SIP-Q gateway to the called SIP telephone
These call legs are seen by Lotus Sametime Unified Telephony as two independent calls, with no correlation between them. The only end-to-end signaling between the originating SIP telephone and the terminating SIP telephone is that which is provided by the PSTN or SIP-Q signaling connection; therefore, there may be a reduction of feature transparency (for example, no calling name display) compared with a call that is wholly routed via the private network.
When rerouting becomes necessary, Lotus Sametime Unified Telephony does the following:
Parent topic: CAC (Call Admission Control) Rerouting
Rerouting a Call to a SIP Subscriber
- Immediately routes all subsequent calls to the next available route.
Marks the unresponsive first gateway as inaccessible, and stops routing calls to it.
Performs an audit of the unresponsive gateway for a provisionable time.
Automatically switches back to the first gateway when the audit mechanism indicates that the gateway has become responsive.