Configuring the Verba Dial-in Recorder Service
Prerequisites - PBX side configuration
Step 1 - Plan directory numbers used for different available features:
- Unattended recorder line: Call is recorded silently, without any notification. The directory number has to be added to the Verba extension list and the PIN-based authentication has to be turned off.
- Voice recorder line: After directory number and/or PIN code based authentication via voice prompts call is recorded with beep notification.
- Voice player line: After directory number or PIN code based authentication user can playback his/her calls.
- Voice portal line: After directory number or PIN code based authentication user can record the current call or playback his/her calls, or playback calls by directory number if access is granted Controlling is done via DTMF - instant voice response.
- Video portal line: After directory number or PIN code based authentication user can record current video call or playback his/her calls (audio, video), or playback calls (audio, video) by directory number if access is granted. Controlling is done via DTMF - instant video response.
- Open recording lines enabled: Incoming calls to the Verba Dial-in Recorder, regardless the actual called number, will be recorded with beep notification. The caller number has to be added to the Verba extension list.
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:
- Voice playback lines
- Voice recorder lines
- Unattended recorder lines
- Video portal lines
- Voice portal lines
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.