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PBX modernization over SIP

Give your legacy PBX a second life: IP telephony, recording, IVR and CRM — via the Alvis gateway in-line on the E1 stream, without replacing the station or costly cards. Below is material for your role: manager, engineer or buyer.

Free 2-month evaluation Material for my role

How it works

Network modernization scheme

Hover a node — we explain what it does. The dots show calls passing through.

E1 / PRI (TDM)SIP / IPinternal phones

Hover a node on the scheme — we will show what each element does.

Compatibility: which PBXs we modernize

Panasonic — TD500, TDA100/100D, TDA200/600, TDE100/200, NCP. PRI cards turnkey (0290), no licenses needed.
LG-Ericsson — LDK100, LDK300, ipLDK. Turnkey setup, we supply the PRIB.
Siemens — HiPath, HiCom. Russian CallerID Name, HiCom expansion over SIP.
Avaya and many other PBXs with an E1 interface.

Not all models are listed — please ask; owners of many other PBXs reach out to us. We handle import substitution, SIP through crypto-routers, replacing flaky TDMoIP. Alvis is officially supported by the manufacturer. We also help with installation, setup and maintenance of the PBXs themselves.

Scenarios

Four modernization directions

Pick the scenario closest to you and expand the details.

1 · Expanding the PBX numbering plan

FXO/FXS card slots are used up, new expansion shelves are unavailable. Both systems start working as one: the familiar numbering is preserved, and the station keeps running.

On the PBX, XXX/XXXX numbers are routed into the E1 PRI trunk; SIP phones and/or FXS gateways are installed at workplaces
On Alvis, routing from Li#1 to these SIP numbers is configured — directly or via an external Asterisk
PBX slots are gradually freed, IP phones are added on the IP-PBX
New services appear: call recording, voicemail-to-email, multi-level IVR

2 · VoIP optimization

Savings on long-distance and international calls. Alvis reroutes long-distance/international traffic to cheaper SIP routes, and the PBX can additionally connect to multi-channel city SIP numbers — direct savings.

3 · Modern features and contact center

"A lot was paid for the PBX, a pity to throw it away", yet management wants an advanced IVR and a contact center. Alvis adds IP-PBX features (Asterisk, Askozia, FreePBX, Elastix, 3CX, cloud) to your PBX: flexible forwarding to PBX extensions, CRM integration. The PBX works as a large, economical FXS/FXO gateway; unneeded modules (DISA) are disabled. Typically Elastix is deployed (as a VM) — recording, IVR and branch consolidation, while the PBX stays an important link without a "revolution".

4 · Replacing costly E1 streams with SIP

Savings on city E1 subscription fees and more simultaneous calls. Reasons vary: an office move where E1 cannot be delivered; unsatisfactory TDMoE quality; the operator migrating clients to SIP. Alvis replaces the E1 link being retired — nothing needs reconfiguring on the PBX side, and long-distance can go through other SIP providers.

By role

Material for your role

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A full-featured PBX with all IP-telephony functions at minimal cost — without replacing the station.

Which business problems it solves

Slots are used up — new staff and remote offices cannot be connected
Native SIP support is missing or licenses are prohibitively expensive
No short dialing between offices with different stations
The engineer who configured the station has left
Long-distance E1 streams are expensive; you want to drop them smoothly

Typical problems of ageing PBXs

Out of capacity — new subscribers cannot be added.
Native SIP support is missing or too expensive.
Connecting an IP phone over the Internet is rather problematic.
You want prefix-free dialing to another office at 2xx, but it has a different PBX and its own admins.
The person who configured the station has left, and the new admins don’t know how to expand the PBX for new business needs.
There is only a built-in DISA with a bug, and on a limited number of channels.
E1 is expensive and many drop it — but you want to do it smoothly so users are never left without a line.

What the company gets

More subscribers via IP phones — no cards or shelves to buy
Lower long-distance costs: remote extensions and cheaper SIP routes
New services: recording, voicemail-to-email, IVR, contact center, CRM (1C, Bitrix, AmoCRM)
A unified numbering plan for HQ and remote offices
A smooth migration without downtime, preserving the PBX investment

Reliability and guarantees

Over 1000 deployed systems in Russia, CIS and Europe, 1500+ clients; used in defense projects
5-year warranty, SSS declaration, Russian manufacturing
After connection — a one-month pilot period with monitoring and a commissioning act

Why E1/PRI integration pays off

E1/PRI integration with a SIP/E1 gateway is the most cost-effective and secure modernization path, fitting all PBX types with E1. GarantPlus helps with setup: in 15 years we have learned everything about E1, we help connect over SIP to operators and IP-PBXs, configure your PBX with our own staff, install a FreePBX / Elastix IP-PBX and train you. The result is a PBX with all IP-telephony features at minimal cost: more subscribers via IP phones and FXS gateways, lower long-distance costs through remote extensions for IP operators, and the ability to add new subscriber services.

Alvis in-line on E1 adds SIP; signaling on an embedded Asterisk, real-time on the RtpBridge engine (DSP).

How it connects and handles calls

To the city — as user, to the PBX — as net, and in-line on an existing E1
Hardware timeslot cross-connect: in-line E1↔E1 transit does not load the CPU — no quality loss, no extra latency
SIP registrar: registers IP phones and SIP trunks, works as a client of operators, cloud and IP-PBXs
If slots are occupied, an FXO/FXS card is removed and its subscribers move to IP

How the modernization process goes

A SIP/E1 Alvis-GW-2E1 gateway is connected over E1. If a card and a city (PSTN) E1 uplink exist, Alvis is inserted in-line on the link — a hardware timeslot switch ensures transit-call quality and no extra latency. The client connects Alvis to the LAN and the E1 streams, and GarantPlus specialists connect remotely via TeamViewer and help configure it — we have 15 years of E1 experience.

If all slots are occupied, one FXO/FXS card is removed and its subscribers move to IP phones or an FXS gateway. GarantPlus can supply E1 PRI cards for common Panasonic / LG PBXs turnkey. Basic configuration of the whole system is included in the equipment price (and available under the free-testing programme): we help configure your PBX, install Elastix or another IP-PBX, and connect to Mango and other cloud services. We have staff specializing in Panasonic / LG / Avaya — we can help from the other side too.

Flexibility and reliability

Flexibility: grow the hybrid PBX’s internal subscribers — add FXS subscribers and any SIP-compatible popular IP phones. Phones can be at the main office or remote ones, keeping a single 3- or 4-digit numbering plan.
If the PBX lacks an E1 PRI card, one is installed; if all slots are taken — in place of one of the FXS-port cards.
Reliability: only proven components, full production control in Russia and Taiwan — we do not manufacture in mainland China. The circuitry is battle-tested over the years, performance is guaranteed. The equipment is used in defense and other special projects, and there are extended-temperature versions.
The client gets a solution, not a bare gateway: over 1000 deployed systems, plus help with services (call-center setup, debtor call campaigns, etc.).

The RtpBridge core

An embedded Asterisk handles SIP and PRI signaling and keeps the extensions.conf dialplan (with macros). Real-time RTP/E1 under full load (30 SIP↔E1 + up to 30 E1↔E1) is processed by the RtpBridge engine on DSP — hence reliability under load and easy interconnect with Asterisk, FreeSWITCH and corporate IP-PBXs.

Signaling, voice, codecs

E1 protocols: ISDN PRI (EDSS1), QSIG, SS7; port as NET or CPE
Russian CallerID Name pass-through (incl. Siemens HiPath)
Hardware echo cancellation LEC G.168+ up to 128 ms
Codecs and transcoding G.729, G.723, G.711, AMR-NB; T.38 fax

Integration and setup

Interconnect with Asterisk-based IP-PBXs: FreePBX, Askozia, Elastix, 3CX, CommuniGate Pro, FreeSWITCH. A simple web interface (approved by Panasonic installers), plus an Online Demo. The client connects Alvis to the LAN and E1 streams; GarantPlus engineers help remotely (TeamViewer / SSH under NDA). SIP-trunk and in-line E1 examples are on the AlvisWiki.

Redundancy, security, monitoring

The Rescue-4E1 module — a FailOver Switch with hot standby and rerouting
SIP protection: fail2ban and others
SNMP monitoring of E1 streams, PRI signaling, CPU, memory and L1/L2/L3 alarms (Zabbix / PRTG)

Basic setup included, no SIP licenses, a free trial with no deposit, a 5-year warranty and an SSS declaration.

Cost and licenses

Basic configuration of the whole system is included in the price — no extra charge
No licenses are required for SIP connections
If needed, we supply E1 PRI cards turnkey (Panasonic 0290, LG PRIB)

Warranty and support

5-year warranty; during the term we resolve all issues, including compatibility
After the warranty — free consultations
For VIP clients — priority support with on-site visits across Russia and CIS

Risk-free free testing

A fully functional unit at your office, delivered, for up to 1 month
No deposit or prepayment — only a guarantee letter
We help with the setup: IP phones, SIP operators, IP-PBX, call-center
We connect to CRM: 1C, Bitrix, AmoCRM

Delivery, certificates, manufacturing

Usually shipped the next day after the guarantee letter
The system ships ready to use
SSS declaration; proven components, manufacturing in Russia and Taiwan
Verified SIP operators: RTK, MTT, SIPNET, VoipBuster and others

Pilot period and commissioning

We don’t abandon the client after purchase and monitor system stability during the pilot period. From the moment of connection, a one-month trial run lasts: by agreement with the customer we run additional monitoring of the system (over SSH with an NDA signed, or via TeamViewer), learn the specifics of local E1 PSTN operators’ settings, determine the required SIP-connection settings and resolve security matters. On completion, a Commissioning Act is signed. Since Asterisk is inside, configuring SIP trunks is not hard, and long-distance and international calls can be routed to popular VoIP SIP operators’ services.

Evaluation

Test the modernization on your PBX — free for up to 1 month

We ship a gateway, help configure E1 and SIP. No deposit — just a guarantee letter.

Free 2-month evaluation Catalog