Keep established PBX users
Retain the extensions and calling habits that still support daily work.
Modernize your PBX without replacing everything at once.
Bridge a traditional PBX to SIP trunks, IP voice systems and new extensions through OpenVox analog, digital and SIP interworking.
The migration decision
Keep what works while the network evolves.
Many organizations still depend on existing extensions, numbering and trunk arrangements even when new sites or SIP services are being introduced.
A coexistence design avoids tying every improvement to one system replacement. OpenVox gateways create controlled analog, digital and SIP paths between the current PBX and the target voice environment.
OpenVox bridge layerAdd new trunks or IP users
Introduce SIP connectivity, IP extensions or another site without forcing an immediate core replacement.
Migration paths
Choose what changes first.
PBX coexistence becomes easier to plan when analog, digital, SIP and expansion paths are separated clearly.
Connect analog PBX lines
Use FXO access where the existing PBX presents analog trunk or PSTN connections.
Preserve E1/T1 interworking
Use digital gateway or DTU capacity for documented E1/T1/J1 and signaling requirements.
Introduce an IP voice core
Use SIP interworking to connect the legacy environment with an IP-PBX, softswitch or IMS network.
Add users or another location
Extend calling without making the first project dependent on replacing every legacy component.
Move from PBX inventory to controlled coexistence.
Document the present system, select the correct interworking path, validate calls and migrate only the elements included in each phase.
Inventory
Record PBX model, trunks, extensions, numbering and required services.
Select interfaces
Match analog, digital or SIP roles to the existing network.
Validate coexistence
Test inbound, outbound, internal and cross-system calling behavior.
Migrate in phases
Move trunks, users or sites according to operational timing.
Combine analog, digital and SIP interworking where required.
Product selection follows the existing PBX interfaces, trunk capacity, target IP system and migration sequence.
DGW-L304
Four-span E1/T1/PRI VoIP gateway for connecting up to four digital trunks to SIP networks with up to 120 concurrent calls.
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iAG200
Compact 2-port FXS gateway for connecting analog phones and fax endpoints to SIP-based voice platforms in small offices, retail sites, and SOHO deployments.
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UCP1600
Unified Communication Platform (UCP) is the ideal solution to integrate data, voice and video in one product for small and medium-sized enterprises, which can be used in scenarios such as government and enterprise call systems, commercial call centers, and dispatch center industries.
View productVerified interfaces for coexistence.
Based on OpenVox UCP, analog gateway and digital trunk documentation.
FXO interworking
OpenVox analog gateways and modules connect analog PSTN lines and PBX interfaces to SIP systems.
Proof chips
- ANALOG
E1/T1/J1 access
Documented digital products provide selectable trunk and signaling options.
Proof chips
- DIGITAL
IP voice interworking
UCP supports standard SIP trunks for IP-PBX, softswitch and IMS environments.
Proof chips
- SIP
Behavior validated before cutover
Numbering, signaling, call routing and service requirements are confirmed during solution design.
Proof chips
- PROJECT FIT
Turn your current PBX into a phased modernization plan.
Share the PBX model, trunk types, extension estate, target SIP system and migration priorities. OpenVox can help define the analog, digital and SIP interworking roles.