What does VoIP trace in a Cisco Unified Border Element do?
Answer(s): D
A company is experiencing an issue where calls from the Cisco UCM via Cisco Unified Border Element to the ITSP fail with a "500 server internal error" right after a 183 session in progress is sent back from the ITSP.What must be enabled to resolve the issue?
Answer(s): B
Refer to the exhibit. Users report that outgoing calls do not work on the new SIP trunk for outgoing calls. The solution consists of a Cisco UCM Cluster linked to a Cisco Unified Border Element where the SIP trunk is terminated. The provider required 10 digits. The logs show a line going toward the Cisco Unified Border Element. Which code snippet must be added to the configuration to meet the requirement?
Answer(s): A
The SIP INVITE log shows that the calling number is being sent as 1155@..., which is not in the 10-digit format required by the ITSP. To rewrite the number in the proper format, such as 9135551155@..., a SIP profile with header modification must be used.Using the sip-header modify command in a SIP profile, applied to the INVITE request, allows you to match and rewrite the header:request Invite sip-header modify "<sip:1(...)@" "<sip:9135551\1@" This ensures that the call sent to the ITSP includes the correct 10-digit number, enabling the call to be accepted and routed properly.
A Cisco engineer troubleshoots an issue with calls through the Cisco Unified Border Element, where outside callers experience one-way audio after a party initiates a transfer. Which command must be configured on the Cisco Unified Border Element to resolve this issue?
Answer(s): C
The midcall-signaling pass-thru media-change command ensures that media renegotiation (such as codec changes or media IP/port updates) is correctly passed through during midcall events like call transfers. Without this configuration, Cisco UBE might not properly handle the re-INVITE or UPDATE messages, leading to one- way audio. This command allows media parameters to be updated correctly during midcall signaling, which resolves issues like one-way audio during transfers.
Refer to the exhibit. During a troubleshooting session, an engineer was presented with this call flow output diagram after a user reported issues with a call via the SIP ITSP to a counterpart. What is occurring with this call flow?
The SIP call flow diagram shows multiple Re-INVITE and ACK messages that include SDP (inactive), which means the media streams are marked as inactive. This status persists throughout the midcall signaling, including after the call is resumed from hold, without ever switching back to SDP (sendrecv).Since both sides are negotiating and acknowledging media as inactive and never transitioning back to active media, this results in no-way audio after the call is resumed.
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