Frequently Asked QuestionsWe are committed to providing courteous, professional and responsive customer care through our team of highly skilled support engineers to help assess and resolve your product issues as quickly as possible, to ensure uninterrupted operation of your network.
Frequently Asked Questions
Yes, Talking SIP's architecture is designed specifically to give the service provider the flexibility to service any point of presence either at a local node level or from a central location without any additional licensing cost.
Talking SIP scales from a starting density of 30 sessions/concurrent callers to thousands of concurrent callers. As many as 2,000 sessions/concurrent callers are supported in a single node/blade. Additional nodes can be easily added to a network, while utilizing a centralized database, in order to scale out the network. Network size and scale is a function of the server hardware in which it is running on. Example Talking SIP call processing and database hardware configurations can be found on our website by clicking here: http://www.ivr.com/talking-sip-server
Each component in the Talking SIP architecture can be easily made redundant. There are many available options ranging from the License Manager level, communications node, to the database level for replication/failover, all the way to clustering for mission critical implementations -- all options are cost effective and readily available.
Talking SIP is an advanced SIP based solution utilizing advanced VoIP signaling in order to provide real-time media, application and billing services to a wide assortment of SIP based devices. By being a pure SIP solution Talking SIP is an intelligent solution that offers real-time call state management, media streaming (in all of the popular codecs) along with broad industry interoperability. RADIUS is a legacy technology that was initially used for dial up modem authentication and is a limiting method of providing AAA services in a VoIP deployment. Some of these limitations include lack of industry support across varying devices, inefficient use of network resources in order to service a call, complicated and limited TCL call scripting and limited and hard to manage TFTP voice prompting.
Yes, Talking SIP comes standard with support for postpaid/wholesale billing including support for real-time credit limit enforcement and account invoicing. Callers can be authenticated by the SIP URI From/To information, an IP address from a peering partner's gateway, carrier code, SIP registration credentials, Remote-Party-ID, DNIS or ANI.
Yes, Talking SIP is commercially deployed in various countries providing primary and secondary line services for residential and business users accessing the network via softphones, IADs and IP phones. This is a popular application for the use of our product in order to provide tandem switching, media streaming of voice prompts and real-time billing for both flat-rate/packaged and per minute subscriber offerings.
Talking SIP comes complete with our Telephony Management Console that allows varying levels of access in order to perform system administration across a global network from a single login session.
Talking SIP provides Class 5 services and optional voice mail and optional three-way conferencing capabilties, and it also interoperable with varying SIP based solutions (premise-based and hosted) allowing the service provider to select best of breed solutions for integration into the overall network, helping to build margins and drive revenue.
Yes, Talking SIP's Telephony Management Console can be provisioned in order to provide very granular level access to users that are enrolled in varied user groups. These user groups contain the selected access rights made available from the Telephony Management Console menu. Those areas of the system where access is not being provided are entirely removed from the menu thus eliminating potential questions from users. Talking SIP has been designed from the ground up to be successfully deployed in hosted and/or self managed models.
Talking SIP interoperates with any RFC 3261 compliant softphone, IAD, gateway, softswitch, IP phone, session border controller (SBC) and/or proxy server. While we continually test with best of breed industry partners and welcome the opportunity to engage with other market leaders as our customers' needs require.
Talking SIP is compliant with SIP RFC 3261 and relies on support for the RFC 2833 and/or SIP INFO methods for DTMF recognition, which is used when the caller will be prompted for DTMF digits for an account number and/or a destination number. Please note that with RFC2833 Talking SIP uses Named Telephone Events with a Payload type of 127, and with SIP INFO uses the Cisco implementation. The other requirement is for support for the SIP REINVITE for two stage dialing applications, to allow the redirection of the media streams directly between the endpoints while Talking SIP continues to maintain full control of the signaling to enable real-time call cut off.
Talking SIP ships standard with over 900 pages of professionally written documentation along with a context sensitive on-line help system that is always available through the management interface by simply hitting the F1 key. There is also a 9 step Quick Installation Guide and computer based training (CBT) videos that walk you through in lock step on the installation and configuration of both the Microsoft SQL server software and Talking SIP.
Talking SIP provides support for voice prompting in unlimited languages with the ability to custom tailor the language selection based on DTMF key press, account, IP address or DNIS. Talking SIP also supports multiple currencies as well as the ability to set the time and date format on a regional basis.
Talking SIP ships standard with the prepaid/postpaid calling card, tandem switching, termination and voucher recharge modules.
Talking SIP is based on the affordable, robust and reliable Microsoft Windows 2012 Server OS and Microsoft SQL Server 2012 database. Talking SIP is a Certified for Microsoft Windows Server 2012 and Microsoft SQL Server 2012 product and has undergone independent third party testing in order to achieve this recognition.
- Intel Pentium IV or Core 2 Duo Processor 2.2 GHz or Higher
- At least 512MB of RAM
- Windows Server 2003 or 2008 higher.
- At least 125MB free disk space
- Client connectivity component from the Microsoft SQL Server 2008 CD (Needed to connect to the SQL Server subscriber database)
- License Manager
- Intel Pentium IV or Core 2 Duo Processor 2.2 GHz or Higher
- At least 1,024 MB of RAM
- Windows Server 2003 or 2008 or higher
- 5 GB disk space
- 100/1000 MBPS LAN card
- Microsoft Windows Networking with TCP/IP installed
- Microsoft SQL Server 2008 or higher software
Talking SIP ships standard with over 127 comprehensive reports. We continually add new reports and can work on a professional services basis to develop additional reports on a customization basis as required by the customer.
There is also an optional report designer that is integrated into the Telephony Management Console to allow existing reports to be modified or wholly new ones created.
No, Talking SIP has been designed to handle report updates automatically eliminating the need to distribute software updates to users. All reports are stored in the SQL database, so when a report is modified or added the changes are automatically and immediately available to all logged in users.
Yes, available as an option Talking SIP supports the ability to process credit card transactions either via the Internet or via the telephone via DTMF key presses.
Yes, available as an option is our End-user Web Interface and Credit Card Recharge modules that allow for the automatic subscription of services via a credit card as well as end user self-management for such tasks as self service subscription, account recharge, reviewing detailed call history reports, debit/credits posting history, speed dial management, ANI assignment, call statistics and viewing of account balances as well as viewing/editing account information.
Upgrades can be performed in a just in time fashion electronically through the included license manager software. Determining when an upgrade is appropriate can be measured via the call statistics screen and/or the license usage report to ensure conformance with service level agreements.
The complete installation and initial configuration of Talking SIP can be achieved in under an hour by a network technician familiar with the Windows OS and Microsoft SQL server software.
Yes, Talking SIP supports the import and export of rate decks in a .csv format allowing for easy interoperation with Microsoft Excel. Once your rate deck is built or modified simply import it into Talking SIP and your changes will immediately take effect.
Yes, Talking SIP provides multiple methods of fraud filtering consisting of white/black lists that can automatically block a caller by their ANI, account or by a range of IP addresses. Blocking can be performed on a temporary or permanent basis. Other fraud prevention features include; up to 40 digit randomized account number generation, second level PIN prompting, single simultaneous use account enforcement (can be toggled on/off at system administrator discretion) as well as informative risk management reports to help run a secure network.
Talking SIP employs a centralized license manager that promotes efficient use of resources across a globally deployed network. High availability and efficient resource utilization are key considerations to the success of any mission critical and distributed network. By centralizing the licensing of Talking SIP, network operators avoid situations where there is blocking occurring at one network point while idle capacity sits at another. Session licenses can be allocated on a first come first serve basis or pre-allocated on a SIP endpoint-by-endpoint basis. As opposed to enforcing licensing within Talking SIP at the network's edge -- this licensing architecture encourages service providers to distribute the processing load across multiple servers for redundancy and performance considerations since there are no financial licensing disincentives to scaling out one's network.