The SIP Credential information that is present in LiveVox PBX is obtained from LVP. This information can be updated to add further details specific to PBX.
- Select the SIP Credential you want to modify on the SIP Credentials window and double-click.
The Edit SIP Profile window appears. - Make the necessary changes in the Basic Details, Caller ID, and Options tabs. Use the information in the following tables for reference.
Basic Details Tab
Field Description Name Enter the SIP user name Extension Enter the number that the SIP user can be reached at, using an SIP phone or deskphone. External Number Enter the number that the SIP user can be reached at, can be a number provided by the company. Domain This field is greyed out and cannot be set by the client. Location Select the site location that this SIP credential is associated with. User Select the LVP user that this SIP credential is associated with. Voicemail Box Select a voicemail box to associate with this credential. For information on adding a voicemail box, see Adding a Voicemail Box. Username This field is auto-populated. Password Use the default password or enter a password manually or click Generate to use a system-generated password.
Changing the password disables any device currently using these credentials. You will have to reconfigure your device accordingly.
Emergency Services Location Address Address 1 Enter a valid, physical address that emergency service personnel may reach the user at. Address 2 City State Zip Caller ID Tab
Field Description Local Calls and Outbound Calls Override Default Caller ID Use the slider to determine the visibility of the default caller ID. Caller ID Name Enter the Caller ID Name for the agent. Caller ID Number Enter the Caller ID number. Options Tab
Field Description Bypass Media Never
Always
Live Transcribe No
Yes
Allow International? No
Yes
Redirects Calls To Enter a valid telephone number Record Calls None
All
Inbound
Outbound
Local
An agent can record their calls only according to the option configured here.
Music On Hold For information on managing files that can be used here, see Adding Hold Music. Call Timeout Duration in seconds that unanswered inbound calls will ring for before being disconnected. Force Codecs OPUS
Siren @ 32Khz
G722.1 @ 32khz
G722.1 @ 16khz
G722
Speex @ 32khz
Speex @ 16khz
G711u / PCMU - 64kbps (North America)
G711a / PCMA - 64kbps (Elsewhere)
G729 - 8kbps (License Required)
GSM
Siren (HD) @ 48kHz
Siren (HD) @ 64kHz
VP8 [Video]
H264 [Video]
H263 [Video]
H261 [Video]
- Click Save SIP Credentials to save the changes.