GEOP-SIP Sample Questions & Answers

The t-invite parameter specifically controls the amount of time SIP Server will wait for a final response (like 200 OK) to an outbound INVITE message sent to an endpoint. In a high-latency scenario, the default value may be too short, causing SIP Server to time out and initiate No Answer Supervision prematurely. Increasing this value gives the remote endpoint more time to respond over the slow link. ringing-timeout applies after a 180 Ringing response is received, and the others relate to different timers.

The gcti-user-data-map option is crucial for multi-site call tracking. It specifies which UserData keys (containing information like call type) should be encoded into a proprietary SIP header (like X-Genesys-GCTI-Data) when SIP Server transfers a call to another site. The receiving SIP Server at the second site then uses the same map to decode the header back into UserData, thus preserving the call context for consistent tracking and reporting across the enterprise.

The 'Barge-in' feature is implemented by creating a three-way conference call between the caller, the agent, and the supervisor. If Media Server is not correctly configured to support conferences, or if conference resources are unavailable, the barge-in attempt will fail. Silent monitoring, on the other hand, typically uses media forking or a passive tap and does not require conference resources, which explains why it would still work.

To block specific SIP headers, you must use two options in conjunction. First, sip-header-filtering-mode must be set to block-list. Second, the sip-headers-to-filter option must be configured with a comma-separated list of the headers to be blocked, such as P-Asserted-Identity,X-Custom-Info. Using only one of these options is insufficient to enable the filtering.

The verbose log level in SIP Server controls the amount of detail in the log files. A setting of 5 is specifically designated for SIP message logging, which directs SIP Server to write the full text of all received and sent SIP messages to the log. This is essential for deep troubleshooting of signaling issues.

A GCTI DN with subtype ServiceInitiatedCall is used to represent a logical, non-physical endpoint that can originate calls. This is essential for applications like outbound dialers, proactive notification systems, or IVR ports that need to place calls into the telephony environment. SIP Server uses these DNs to manage the signaling for calls that are not initiated by a human agent at a physical phone.

In the Network SIP Server architecture, proxy instances are deployed at local sites to serve as the CTI connection point for local applications like Universal Routing Server (URS). The proxy receives T-Library requests from the local URS, forwards them to the central control SIP Server for processing, and passes responses back. This model centralizes call control logic while distributing the CTI load and improving resiliency.

The correct approach involves two steps. First, configure SIP Server to map the value of the X-Customer-Tier SIP header into a key in the call's UserData (e.g., customer_tier=VIP). Second, in the URS routing strategy, read this UserData key. If the value is VIP, the strategy can use the SetPriority function to elevate the call's priority in the queue, ensuring it is answered before standard calls. This provides a flexible and powerful way to use external data for routing decisions.

False. Stream Manager is specifically required for streaming media from a web server (HTTP streaming). For basic treatments like playing locally stored .wav or .vox files for Music on Hold or announcements, the Media Server can access these files directly from its local or a network file system without the need for Stream Manager.

The best practice is to leverage SIP Server's flexible codec configuration. By creating different codec profiles (or configuring codecs on different DN/Trunk objects), an administrator can define a preferred codec order. For local agents (on a specific switch or trunk group), G.711 can be prioritized. For remote agents, G.729 can be the priority. SIP Server will then negotiate the most appropriate codec based on the destination of the call, optimizing both quality and bandwidth without forcing a single standard on all users.