Configuring the Verba Dial-in Recorder Service

Prerequisites - PBX side configuration

Step 1 - Plan directory numbers used for different available features:

Step 2 - Create trunk pointing to the address where recorder is planned to listen

Step 3 - Create route patterns for dedicated directory numbers

See PBX specific configuration checklists here: Configuring Cisco UCM for dial-in recording, Configuring Microsoft Lync for dial-in recording, Configuring Polycom RMX for conference recording

Configuring the Verba Dial-in Recorder Service

Step 1 - In the Verba Web Interface go to System > Servers > Select your Recording (or Single) Server > Click on the Service Activation tab.

Step 2 - Activate the Verba Dial-in Recorder Service by clicking on the

icon.

Step 3 - Click on the Change Configuration Settings tab.

Step 4 - Expand the Dial-in Recorder node.

Step 5 - Under the Lines node, set the Enable open recording lines setting to Yes if required. Provide the line numbers at the following settings, based on your requirements:

Step 6 - Set the Internal Domain, Numbers Pattern setting. This has to be a regex which matches to all internal numbers.

Step 7 - Save the changes by clicking on the

icon.

Step 8 - A notification banner will appear on the top. Click on the click here link, so you will be redirected to the Configuration Tasks tab. Click on the Execute button in order to execute the changes.

Step 9 - Click on the Service Control tab.

Step 10 - Start the Verba Dial-in Recorder Service by clicking on the

icon.

Assign users to recorder lines

First of all, all users have to be added to the Verba user list, and their line numbers and SIP URIs has to be added to the Verba extension list for enabling for them using the recorder lines. All users have to have a user role which contains the Dial-in interface right under the Application Access section. This can be done also by Active Directory Synchronization.

Once the users and their extensions are present in Verba, the PIN code related settings can be set. If the PIN-based authentication is required, then a PIN code has to be set for every user. To do that, go to the Users \ Users menu, select the user from the list, then set the Recorder Line PIN setting. If the PIN-based authentication is not required, then go to the Users \ Extension menu, select the extension from the list, and turn on the Do not request PIN on Recorder Line setting under the Dial-in Recorder Specific Settings section.

Configuration reference

Recording line settings

Default voice prompt language: voice prompt language for unauthenticated or users where language is not specified
Enable open recording lines: if enabled all calls going to unspecified directory number will be recorded without any authentication
Voice and video prompt's directory: directory for IVR prompts. For customization see xxxxxx
User response timeout: call will be timed out and terminated if there is no user response for requested action until this time
Different feature lines: one or multiply numbers where given feature will invoked.

SIP settings

Call timeout in sec: SIP session timer, if call keepalive fails call is terminated and considered timed out
RTCP support: support for Real-Time Control Protocol, based on this network/bandwidth adaptation for encoders/decoders is possible
SIP r-port: support for SIP symmetric response routing (RFC 3581)
Force duplex streams: the recorder can act as receive only endpoint according to SIP/SDP negotiation, however some devices do not honor this, and terminates the call because of media timeout. If duplex media is forced recorder acts as send-receive endpoint, and generates media. If it is not forced most of the MCUs hide the recorder in the conference, so from video conference recording point of view we would recommend disable it.
SIP user, password, uri for registration, register as client: if trunk based integration with PBX is not prefered, the recorder can register as user agent, however in this case it can serve only one directory number. SIP address is registration uri config, user name is the user used for digest authentication
Recorder display name: SIP display name of the recorder
RTP port range begin - end: RTP port range used by the recorder
SIP signaling transport: prefered transport for recorder initiated SIP sessions
Local SIP port: SIP port on which the recorder is listening. Be sure that configured IP address and local SIP port match the trunk destination address in the PBX

Recording settings

Automatic Gain Control: enables AGC on voice streams
Verba API port: API port for internal service management
Voice call recording format: storage format for audio only calls
Database cache directory: database cache file path
Endpoint emulation:endpoint profile, the followings are supported currently:
- Basic Audio: audio only endpoint with G.722.1, G.722, G.729, G.711 and GSM support
- Basic Video: audio and video endpoint with G.722.1, G.722, G.729, G.711 and GSM, H.264 (SQCIF - 1080p) support
- SIPREC single stream: SRS: SIPREC based endpoint, calls with SIPREC content will be always recorded, it overrides line settings. Single stream media is forced
- SIPREC dual stream: SRS: SIPREC based endpoint, calls with SIPREC content will be always recorded, it overrides line settings. Dual stream media is preferd, but SRC might negotiate in single stream
- Different Cisco Telepresence endpoints: TIPv7.1 based interoperability with Cisco Telepresence. It is still under development, only for experimental use.
Recorder API port: controling port, which makes possible starting outgoing calls from the recorder to playback, and/or record the call
Video call recording format: storage format for video calls
Write XML metadate: write CDR XML with the calls

Advanced settings

Strip domain part of SIP phone number: keep only the user part of SIP uri
RTP stream reorder buffer length: audio reorder buffer size
Media format fallback enabled: in case of not supported codecs, too many streams, not supported streams, transcoding quality issue, the recorder can inteligently change storage format to different kind of codecs which might preserve the recording in more optimal quality.
Always negotiate single codec: in case of SDP offer the recorder will select one codec in each media stream's codec list in the answer. We support handling of list of codecs, and dynamic codec changes, so only in case of interoperability issue should this be enabled.

Configuration Parameter Name	Description

Configuration Parameter Name	Description
Local SIP Port Number	Port number used for SIP signaling communication. This port number has to match the configuration in other systems connecting to the service.
Silent Monitoring Enabled	Enable silent monitoring capabilities for the service. By enabling this option, any ongoing call recorded by any recording service supporting silentmonitroing,can be monitored through the phone playback access numbers. This option does notenables/disables the silent monitoring feature on the web application for calls recorded by this service.
Called Party Name	Display name of the service, which will be displayed as called party name on the caller phone device.
Default Menu Language	Default language setting for the voice menu.
SIP Transport Protocol	SIP signaling transport protocol configured in the SIP proxy. Values can be TCP or UDP. TCP is recommended.
Audio Format	The recorder application will use the selected file format and codec option to create the audio files.
PIN Entry Retries	Number of allowed PIN code entry attempts. After exceeding this number, the system plays an error prompt and disconnects the caller.
Public Recorder Access Number	Entry point (directory number) used to access the recording functionality. Using this access number, the system does notauthenticatesthe caller and allows to access the recording service from any phonenmuber. After connecting the caller, the system plays in a prompt to notify the parties in the call about the callrecording,and automatically starts the recording. If you do not want to allow unauthenticated access to the recording service, do not configure this access number, leave it empty.
Authenticated Recorder Access Number	Entry point (directory number) used to access the recording functionality. The system authenticates and identifies the caller based on the calling party phone number and optionally the user also has to enter a PIN code. If the calling party phone number cannot be found, the user has to enter the PIN code. After authentication, the system plays in a prompt to notify the parties in the call about the callrecording,and automatically starts the recording.
Authenticated Recorder Access Number without Prompt	Entry point (directory number) used to access the recording functionality. The system authenticates and identifies the caller based on the calling party phone number and optionally the user also has to enter a PIN code. If the calling party phone number cannot be found, the user has to enter the PIN code. After authentication, the system DOES NOT play in any prompt, it automatically starts the recording.
Authenticated and Unauthorized Playback Access Number	Entry point (directory number) used to access the playback functionality. The system authenticates and identifies the caller based on the calling party phone number and optionally the user also has to enter a PIN code. If the calling party phone number cannot be found, the user has to enter the PIN code. After authentication, the system asks for a phone number, which is used to query the database and offer calls for playback or silent monitoring. Silent monitoring is only available for the ongoing calls. It is important to understand, that using this access number, the system does not check any authorization to access certain calls or group membership right or information while offering calls for playback or silent monitoring. The systemsimpleoffers all calls. If you do not want to offer unauthorized access to the calls, do not configure this access number, leave it empty. The playback functionality is available for all voice calls recorded by anyrecordinhservice. It is not limited to the calls recorded by the Dial-in recording service.
Authenticated and Authorized Playback Access Number	Entry point (directory number) used to access the playback functionality. The system authenticates and identifies the caller based on the calling party phone number and optionally the user also has to enter a PIN code. If the calling party phone number cannot be found, the user has to enter the PIN code. After authentication, the system asks for a phone number, which is used to query the database and offer calls for playback or silent monitoring. Silent monitoring is only available for the ongoing calls. The system automatically checks the user authorization to access the calls and only offers those calls for playback or silent monitoring, which are available for the user based on her/his settings. The playback functionality is available for all voice calls recorded by any recording service. It is not limited to the calls recorded by the Dial-in recording service.

Advanced settings

The following table provides detailed instructions on each configuration setting:

Configuration Parameter Name	Description

Configuration Parameter Name	Description
Process Unicode Names	If enabled, the system treats the calling party names asunicodecharacters.
Store Call Time in UTC/GMT	Enables the use of UTC time when writing to the database or XML CDRs. The Verba Web Application treats every date and time value as UTC, so if this setting is turned off, it may resulttodisplaying inaccurate date and time values.
Strip Domain from SIP Phone Numbers	If enabled, the system automatically strips the domain information from the SIP addresses and leaves the phone number information only.
Menu Timeout (seconds)	The system wait that long in seconds for a DTMF input from the user. If the timeout expires and the user did not enter any code, the system automatically disconnects the call after a warning prompt.
Call Timeout (seconds)	Defines the call timeout value in seconds, which is used to terminate the call if the reinvite was not successful.
Database Cache Folder	The path to the database cache file withoutfilename. Network drives are not supported, because of reliability and performance issues, so please do not use mapped network drives or UNC network drives, use only local folders. Use the browse button to select the proper folder.
Automatic Gain Control Enabled	If this setting is enabled, the application automatically controls the gain in the audio file to provide more convenient userexperiancewhile listening back recordings.
IVR Prompt Path	The path to the folder containing the prompt files used by the IVR. Network drives are not supported, because of reliability and performance issues, so please do not use mapped network drives or UNC network drives, use only local folders. Use the browse button to select the proper folder.