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PBX IPS 416 User Manual

The IPS phone system is a 4 telephone line and 16 extension to Analogtelephone system designed for use in a small office and the home office.The IPS by BBS Telecom is simple-to-use, quick-to-install, easy-to-configure SOHO PABXsystem. You can download both the IPS and Plexus telephone system and user guidesin PDF. You will find thePlexusMotel Telephone System manuals on our Plexus page.

PBX IPS 416 User Manual

The difference goes back to the history of telephone systems. Originally, a PBX required switchboard operators to connect internal callers to other lines. Operators did this manually by literally unplugging lines and plugging them into the right extension.

This section is used to define the email address and server the PBX will use to send out Unified Messaging and notification emails. Unified Messaging allows a user to receive emails whenever they receive new voicemail messages. If the PC where you check your email has the capacity to play .wav files, the you will be notified when an email is received, and will be able to listen to the new message directly from the email. Notification emails are sent out for features like Log Watch/Ban.

Many times the email account provided by the end user won't work, or is giving difficulty. Its been our experience its best to not waste time trying to figure out something you don't have control over. In these cases, its advisable to create a gmail account and use that for Unified Messaging. Below is an overview of how this would be configured.

If you wish to manually connect to a specific geographic edge location that is closest tothe location of your communications infrastructure, you may do so by pointingyour communications infrastructure to any of the following localizedTermination SIP URIs:

Twilio will automatically populate the user part of the SIP URI based on theTwilio number the call from the PSTN is destined towards. For example, if thecall from the PSTN is received for Twilio number +14158675309, which isassociated with this trunk, the resulting URI sent to your communicationsinfrastructure will be:

Alternatively, you may also configure a specific user-part (e.g. "anniebp")within the origination SIP URI. Note that the same URI will be used for allNumbers associated with this trunk. Hence, if the call from the PSTN isreceived for Twilio number +14158675309, which is associated with this trunk,the resulting URI towards your communications infrastructure will still be thefollowing for all phone numbers:

The P-Asserted-Identity header contains the phone number of the user who is billed for the call. If Privacy:id is set, this indicates that the information in the header has to be hidden from the call recipient.

If the connection between the Teams client and the SBC is working correctly, but some users cannot make calls, the issue might be caused by incorrect settings or incorrect provisioning of those users.

Businesses do not have to interfere with their external communication operations with the IP PBX system. When the IP PBX is turned on, businesses can keep their original phone numbers. The IP PBX converts any local call over to the network inside the business and permits every user to be able to share the same external lines.

The IPPBX phone system has one or more SIP telephones, a server and an optional voice over Internet Protocol Gateway to connect the existing PSTN lines. The server functions similarly to proxy servers. SIP users who use SIP phones or analog phones are able to register with the server, and when a user makes a telephone call they have the IP PBX to do it for them. This server has a list of all the phones and users with their SIP address and then is able to connect internal calls to external calls by a gateway or the voice over Internet Protocol service provider. More information can be found in the FAQ article.

The IP PBX system is able to be managed through the web configured interface, as well as a GUI. This allows the user to easily maintain their phone system. Stand alone phone systems usually have difficult interfaces that are usually managed my technicians.

An IP PBX is easily adaptable for using a voice over Internet Protocol provider when calling long distance or internationally. The cost savings are significant for the user. If a business has different branches, one can call the system in between branches and have free telephone calls.

A business is able to use the IP PBX to have better customer service and productivity. The Internet Protocol phone system is based on the computer so now the user is able to integrate functions with business apps. An example would be when the user has to bring up the callers record and the record will be automatically pulled up when the person calls in, which cuts down searching time. Calls placed out of the business can be taken from Outlook, eliminating the user to type in the number.

If an employee wants to roam, this is for users to be able to work at home. The employee can use the SIP phone and take calls from their extension, as like they do in the office. Calls are also able to be diverted due the SIPs characteristics.

Purchasing an IP PBX software system is good for new companies as well as existing companies who already own a PBX. The IP PBX provides the user with significant cost savings when managing the system, as well as in maintenance and ongoing costs that this choice is the best choice for every company.

When a variety of logical functional options may be assigned to a single cable, then once the cable has been connected between two objects, a configuration window is automatically displayed that allows the user to select the required function for that cable.

The Trash Bin allows the user to delete objects that have been previously placed on the workspace. To delete an object, drag that object towards the or icon at the bottom right-hand corner of the workspace. This icon will be replaced by the Trash Bin icon , and the object to be deleted should be dragged and dropped over the Trash Bin.

The Internal Number object is used for internal dialing as an extension number. This facility allows users to call each other directly and to reach selected objects (such as Conference and Voice Menu objects) via internally-assigned extension numbers. Note that one Internal Number object must be created for each internal extension required, and extension numbers cannot be duplicated on multiple Internal Number objects.

The SIP Account contact method (and also the External Line contact method if applicable) provides users with the option of enabling call recording, including the ability to define the recording direction (inbound and/or outbound) and to record internal and/or external calls. In addition, a "record on demand" feature is available, where the user may dial a predefined feature code to activate call recording.

A caller ID name - The caller ID name, which will be displayed on end-user devices that support this feature. By default, this name is the same as the actual contact name to which this contact method is attributed.

It is possible that call recordings sent to an email address may not be delivered due to file size limitations applicable to the destination email server. Note that if the destination server rejects the file, then that file will be unrecoverable. The approximate file size of call recordings is 7 MB per hour, and users should carefully consider this parameter when selecting a file Delivery Method.

View - Defines how the results are to be displayed. The "Daily" option groups results by day, "Hourly" groups results by the hour, and "Grouped" selects results by the exact hour of the day, performing a peak-time analysis and allowing users to determine at which time of the day most calls were received or made.

View - Defines how the results are to be displayed. The "Daily" option groups results by day, "Hourly" groups results by the hour, and "Grouped" selects results by an exact hour of the day, performing a peak-time analysis and allowing users to determine at which time of the day most calls were received or made.

Alternatively if a transfer (xfer) button is available on the phone device or softphone, answer the incoming call, put that call on hold, call the required Internal Number (extension) on a second line, speak with the person to whom the call is to be transferred, press the transfer button, and the call will be immediately transferred. Note that the call may have to be manually disconnected from the first line if your device does not do this automatically.

Record on demand - Applicable to the SIP Account (also to External Line) contact method, and allows users to initiate the recording of calls in real time. In order to use this feature code, the SIP Account must have the Record on demand option enabled in the Call Recording settings, together with a configured delivery method for receiving the contents of the recorded call.

The CLI Rules are used to automatically modify the source and destination numbers sent to the termination gateway, allowing users to add and remove prefixes or completely rewrite the numbers on a per-gateway basis.

The CLI Rules are available for the modification of source and destination numbers, allowing users to include rewrite rules such as adding prefixes to numbers or deleting country codes from numbers on a per-route basis.


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