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What are two possible solutions?
Cisco SIP IP telephony is implemented on two floors of your company. Afterward, users report intermittent voice issues in calls established between floors. All calls are established, and sometimes they work well, but sometimes there is one-way audio or no audio. You determine that there is a firewall between the...
To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?
End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?A ....
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?A . The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.B . Cisco Unified Communications Manager invoked media termination point resources.C...
What should the administrator configure in the Cisco Unified Border Element to fix this issue?
Refer to the exhibit. An administrator is troubleshooting a problem in which some outbound calls from an internal network to the Internet telephony service provider are not getting connected, but some others connect successfully. The firewall team found that some call attempts on port 5060 came from an unrecognized IP...
How does the customer ensure reachability to ITSP, so that if one device on ITSP fails, the calls will be routed to another device?
A customer is using a SIP trunk to route calls to ITSP to decrease the possibility of downtime, the customer invested in a failover device. How does the customer ensure reachability to ITSP, so that if one device on ITSP fails, the calls will be routed to another device?A ....
Where can you find the RTP IP and port information for both sides?
The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses H.323 protocol (slow-start mode). The administrator requests that you provide the IP and port information of the Real-Time Transport Protocol traffic that had the one-way audio call. You gather the H.225 and H.245 messages...
Which configuration fixes this problem?
After configuring a Cisco CallManager Express with Cisco Unity Express, inbound calls from the PSTN SIP trunk receive a ring tone for 20 seconds and then a busy signal instead of voicemail. Which configuration fixes this problem?A . Router(config)# voice service voip Router(conf-voi-serv)#allow-connections h323 to h323B . Router(config)#dial-peer voice 2...
Which two scenarios are correct?
Refer to the exhibit. In an active SIP call between phone user A and phone user B, phone A initiates a call transfer to phone user C. Which two scenarios are correct? (Choose two.)A . Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in...
For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?
For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?A . interworking between an OOB method and RFC2833 for flow-around callsB . interworking between h245-signal and rtp-nteC . interworking between an OOB method and RFC2833 for flow-through callsD . interworking between h245-alpha...
What is the cause of this issue?
A user in location X dials an extension at location Y. The call travels through a QoS-enabled WAN network, but the user experiences choppy or clipped audio. What is the cause of this issue?A . missing Call Admission ControlB . codec mismatchC . ptime mismatchD . phone class of service...
