Practical Cisco 400-051 Testing Engine

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On-demand recording of active agent calls, available in Enhanced and Premium versions, improves customer service and encourages appropriate and consistent agent behavior and it is a feature of Cisco Agent Desktop.

During the establishment, maintenance and termination of a SIP session, signaling messages are exchanged between the two SIP endpoints. There are two different kinds of signaling “conversations” that those messages take part in: transactions and dialogs.
A transaction is a SIP message exchange between two user-agents that starts with a request and ends with its final response (it can also contain zero or more provisional responses in between). For example, during the termination of a SIP session, one user releases the call by sending a BYE request and the other party replies back with a 200 OK response. This message exchange is called a transaction.
But what happens in the case of the INVITE request? The establishment of a SIP session starts basically with an INVITE request and is considered as completed upon the receipt of the ACK. In this case, the transaction starts with the INVITE request and ends with the 200 OK, so the ACK is not part of the transaction. The ACK can be considered as a transaction on its own. However, when the final response to an INVITE is not a 2xx response, then the ACK is considered as part of the transaction. A dialog is a complete exchange of SIP messages between two user-agents. That means that transactions are actually parts of a dialog. For example, in the case of a SIP session establishment, a dialog starts with the INVITE-200 OK transaction, continues with the ACK and ends with the BYE-200 OK transaction.
The picture below depicts the dialog and transactions that take place during the establishment of a SIP session:

Note: There can also be subsequent requests that belong to the same dialog, such as a BYE or a re-INVITE message. As out-of-dialog requests are considered messages such as an initial INVITE request for a new session or an OPTIONS message for checking capabilities.
There are different SIP headers/parameters that identify the dialogs and transactions, and they will be analyzed in later posts.
Reference: https://telconotes.wordpress.com/2013/03/13/sip-transactions-vs-dialogs/

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Cisco Unified IP Phones use the option 150 value as the TFTP server IP address when Alternate TFTP option is set to No. You can assign only IP addresses as Option 150 values. A maximum of two IP addresses get used, and only the first two IP addresses that the DHCP server provides get accepted.
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmsys/CUCM_BK_CD2F83FA_00

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Using the AFG will allow you to get this license mac before even touching the server. It is provided after filling in the main form of the AFG but it can also be found by looking at the last few lines of your platformconfig.xml file.
Once you have the xml files, you will need to map those to the floppy drive of the VM (no usb support on the VM OVA). There are many ways to do this. I simply use a freeware virtual floppy app that I drop the platformconfig.xml file on and then copy the*.flp image out to the datastore. I’ll end up with a directory on my datastore called AFG that has the host named *.flp images that I will use during install. It also serves as archival of these files in the event the server needs to be re-imaged. This is important because the license mac will change if every parameter is not entered exactly as it was prior. If the license mac changes, you will have to go through the process of requesting new license files to be generated.
Reference:http://angryciscoguy.com/jello/cisco-answer-file-generator-to-the-rescue/

1xx—Provisional Responses 100 Trying
Extended search being performed may take a significant time so a forking proxy must send a 100 Trying response.
180 Ringing
Destination user agent received INVITE, and is alerting user of call.
181 Call is Being Forwarded
Servers can optionally send this response to indicate a call is being forwarded.[1]:§21.1.3
182 Queued
Indicates that the destination was temporarily unavailable, so the server has queued the call until the destination is available. A server may send multiple 182 responses to update progress of the queue.
183 Session in Progress
This response may be used to send extra information for a call which is still being set up.
199 Early Dialog Terminated
Can be used by User Agent Server to indicate to upstream SIP entities (including the User Agent Client (UAC)) that an early dialog has been terminated.
Reference:
http://en.wikipedia.org/wiki/List_of_SIP_response_codes

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You can configure the scope of a directory handler to define the objects that callers who reach the directory handler can find or hear. For phone directory handlers, you can set the scope to the entire server, to a particular class of service, to a system distribution list, or to a search space (either inherited from the call or specified for the directory handler). For voice-enabled directory handlers, you can set the scope to the entire server or to a search space (either inherited from the call or specified for the directory handler).
When callers search a directory handler for a particular name, if the scope of the directory handler is set to a search space, Cisco Unity Connection searches each partition in the search space and returns a list of all of the objects that match the name.
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/8x/administration/guide/8xcucsagx/8xcucsag

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A Re-INVITE, it comes after a session has been established. This means that it will apply to an existing INVITE
after a final response has been received and an ACK has been sent.
You send an UPDATE message prior to session establishment, but that’s an article for another day.
A re-INVITE will have the same Call-ID and From tag as the INVITE it is modifying. It can change every
other header as well as the message body, but those two things tell the SIP stack that this is not a new INVITE. The most common use for re-INVITE is call hold. The party putting the call on hold sends a re-INVITE with SDP indicating that media will no longer be sent. That same party will take the call off hold by sending another re-INVITE with SDP indicating that media transmission will resume.
To demonstrate this, I placed a call to my desk telephone, answered it, started up the Avaya traceSM utility, put the call on hold, stopped traceSM, and then took a few screen shots of the resultant call flow. https://andrewjprokop.wordpress.com/2015/02/10/understanding-sip-re-invite/

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Asterisk (*) matches one or more characters. The * gets processed as a wildcard character. You can override this by preceding the * with a backward slash (\\) escape sequence, which results in the sequence \\*. The phone automatically strips the \\, so it does not appear in the outgoing dial string. When * is received as a dial digit, it gets matched by the wildcard characters * and period (.).

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If a line member does not answer a queue-enabled call, that line member is logged off the hunt group only if the setting "Automatically Logout Hunt Member on No Answer" is selected on the line group page.
Reference: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_1_1/ccmcfg/CUCM_BK_A34970C5_00_ n-guide-91/CUCM_BK_A34970C5_00_admin-guide-91_chapter_0100011.html

Each T1 CAS has 24 channels that can transmit 8 bits per channel each. This gives us
a total of 192 bits. The T1 has one additional bit for framing, bringing the total to 193 bits. Two types of line coding can be used on a T1 CAS. The first type of line coding is called Super Frame (SF). This is an older and less - efficient type of framing. Super Frame bundles 12 of these 193 - bit frames together for transport. It then uses the even – numbered frames as signaling bits. The T1 CAS signaling then looks at every sixth frame for signaling information. This comes out to be 2 bits that are referred to as the A and B bits, which reside in frames 6 and 12.

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Cisco Unified Communications Manager supports dialing using directory URIs for call addressing. Directory URIs look like email addresses and follow the username@host format where the host portion is an IPv4 address or a fully qualified domain name. A directory URI is a uniform resource identifier, a string of characters that can be used to identify a directory number. If that directory number is assigned to a phone, Cisco Unified Communications Manager can route calls to that phone using the directory URI. URI dialing is available for SIP and SCCP endpoints that support directory URIs.