[edsanimate_start entry_animation_type= “shake” entry_delay= “0” entry_duration= “3.5” entry_timing= “linear” exit_animation_type= “” exit_delay= “” exit_duration= “” exit_timing= “” animation_repeat= “1” keep= “yes” animate_on= “load” scroll_offset= “” custom_css_class= “”] Save Money[edsanimate_end]
SIP Trunks
[edsanimate_start entry_animation_type= “shake” entry_delay= “2” entry_duration= “3.5” entry_timing= “linear” exit_animation_type= “” exit_delay= “” exit_duration= “” exit_timing= “” animation_repeat= “1” keep= “yes” animate_on= “load” scroll_offset= “” custom_css_class= “”] Save Time[edsanimate_end]
[edsanimate_start entry_animation_type= “fadeInUp” entry_delay= “1” entry_duration= “4.5” entry_timing= “cubic-bezier(0.39, 0.575, 0.565, 1)” exit_animation_type= “” exit_delay= “” exit_duration= “” exit_timing= “” animation_repeat= “1” keep= “yes” animate_on= “load” scroll_offset= “” custom_css_class= “”]You should use SIP Trunking instead of traditional telephone lines.
Save Substantial money every month! [edsanimate_end]
[edsanimate_start entry_animation_type= “slideInRight” entry_delay= “3.5” entry_duration= “3” entry_timing= “linear” exit_animation_type= “” exit_delay= “” exit_duration= “” exit_timing= “” animation_repeat= “1” keep= “yes” animate_on= “load” scroll_offset= “” custom_css_class= “”]Installation is as fast as Two Days.[edsanimate_end]
[icon name=icon-lightbulb] So how does and existing non-IPPBX work with today’s SIP Trunks?
Most new PBXs are already capable of handling SIP trunks, however in this case we are dealing with an existing PBX that is still working fine and the company does not want to replace it yet. It won’t be required anything other than configuring the traditional Trunk on the PBX; the rest is all transparent for it. There, the call is converted to SIP and sent over the IP network to the SIP trunk provider which is in charge of terminating the call on the PSTN. When an user wants to make an outbound call, it will be first handled by the legacy PBX that routes the call to the trunks connected to the Voip gateway also located at the company premises. This VoIP gateway will convert the analog or TDM signal to IP signal. This signal will flow over the IP network, usually the public Internet, to the SIP trunk provider’s network which can route and terminate calls directed to any PSTN number. Inbound calls follow the exact opposite direction.
[icon name=icon-phone] Get Sip Trunking for your system on one of the Nation’s largest networks for a low as $24.95 per month. Ask Callbox Systems for information on a Voip Gateway for your existing PBX.
SIP Trunking provides all the standard trunk features users have come to expect from traditional carriers with the reliability and redundancy of a multi-carrier MPLS network and powerful call management features that provide business continuity and disaster recovery.
Benefits of SIP Trunking SIP Trunking offers valuable benefits including:
- Reduced Costs Consolidate local lines and trunks on OneStream’s network to reduce hardware costs, reduce trunk costs and to eliminate multiple vendor purchasing.
- Improved Continuity Avoid costly outages with network and geographic redundancy options and pre-defined contingency routing.
- Investment Protection Reuse existing legacy PBX line/trunk cards or leverage IP-PBX native-SIP trunk connectivity.
Today’s IP Phones rely on SIP Trunking instead of traditional telephone lines.
Since you are making the investment in new IP Phones you should look to deploy SIP Trunking to achieve the maximum cost savings and advantages of new IP Phone service. CALLBOX Systems’ phones are compatible with the vast majority of SIP Trunk and Hosted Voice Providers. Callbox Systems recommends Call 7 Networks, one of the Oldest providers of SIP Trunking and Hosted Voice in the World. Callbox System premise based IPBX will support both SIP Trunking as well as traditional PSTN Lines and T-1 Service. Information on SIP Trunking SIP Trunking, from Call 7 Networks, uses Session Initiation Protocol signaling and a native IP-based facility to manage all voice traffic between a customer’s IP PBX platform, MPLS-based global private network, and the Public Switched Telephone Network (PSTN).
SIP trunking is a Voice over Internet Protocol (VoIP) and streaming media service based on the Session Initiation Protocol (SIP)[1] by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and Unified Communications facilities.[2] Most Unified Communications software applications provide voice, video, and other streaming media applications such as desktop sharing, web conferencing, and shared whiteboard.[3]
For more information on SIP Trunking please visit:
https://en.wikipedia.org/wiki/SIP_trunking
Need More Information on 4G Technology: Call 1-800-486-4554
