GarantPlus ITGRIX
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Alvis → FreePBX → Bitrix24

Modernize your PBX,
without scrapping it

The Alvis gateway is installed in-line on the city E1 trunk — between the carrier and the PBX. Call logic moves to FreePBX, while the legacy PBX stays in the rack as a large, reliable analog gateway for its own phones. The result: call recording, queues, remote employees and full integration with Bitrix24 thanks to the ITGRIX connector.

GarantPlus ITGRIX Bitrix for your PBX GarantPlus is a certified partner of Itegro (ITGRIX)
  • 15 years with E1 trunks
  • Over 1,500 clients
  • Manufactured in Russia

Works with Panasonic TDA/TDE/NCP (TD500, TDA100/200/600, TDE100/200), LG LDK100/LDK300/ipLDK, Avaya IP Office, Ericsson MD110 and BP250, Siemens HiPath/HiCom, Samsung OS7020, Iskratel Si2000/Si3000, Meridian M-200, Multicom and Minicom DX-500, Omega, Proton-SSS, MXM500P/MP80P/MP48P — and with other exchanges that have, or can have, an E1 trunk installed.

Diagnosis

What’s wrong with your legacy PBX

The PBX runs — and that’s the only argument in its favor. Everything else is already costing you money; those costs just aren’t tallied in one line.

Calls aren’t stored anywhere

A dispute with a customer can’t be settled, there’s nothing to train new hires on, and manager performance is judged by whatever the boss happens to overhear.

The cost: lost deals and repeated disputes

No CRM integration

A rep picks up without knowing who’s calling. The history gets typed in by hand — or not at all. A missed call vanishes without a trace.

The cost: 10–30% of inbound inquiries never reach a deal

End-of-life, no spare parts

Boards get hunted down on classified-ad sites, and the one engineer who still remembers it is the only one in town. A power-supply failure means days of downtime, not hours.

The cost: the company’s phone system hangs on one person and one board

Every port costs vendor money

Adding 16 extensions means a fee, a license, and installation. Voicemail, conferencing, recording — each a separate activation key.

The cost: hundreds of thousands of rubles for features FreePBX has out of the box

Remote staff can’t be connected

An extension lives only inside the office. A branch, a warehouse, home, a business trip — that’s already the employee’s personal cell and lost oversight.

The cost: some of the company’s calls bypass it entirely

Reporting stops at “how many minutes”

How many calls were lost over lunch, who isn’t picking up, the real load on a department — the PBX can’t answer any of it.

The cost: staffing and headcount decisions are made blind

How it works

Two new elements — and the PBX gets a second life

The Alvis gateway is inserted in-line on the city’s E1 trunk — between the carrier and the PBX. It decides which calls go to FreePBX and which go to the legacy PBX, just as before. We don’t touch the carrier, the cross-connect, or the phones.

Carrier network, PSTN PSTN numbers E1 Alvis-GW-2E1 inserted in-line on trunk up to 60 channels EDSS1 / QSIG / SS7 routing and failover SIP FreePBX IVR, queues recording, reports softphone, remote work no per-port licenses AMI / ARI ITGRIX Connector Asterisk ↔ Bitrix24 events, click-to-call links to recordings audio stays on-premise REST API card, lead, recording in the deal E1 PRI Legacy PBX Panasonic / Avaya / Siemens stays in service Analog and system phones — no replacement not touched deployment zone — new hardware and configuration the PBX moves behind the gateway and runs as a large analog gateway only call data goes to the cloud — there’s no voice loop through Bitrix24

The city E1 trunk enters not the PBX but the Alvis gateway — which is inserted in-line on the link between the carrier and the legacy PBX. The gateway then distributes calls: some go over SIP to FreePBX, the rest pass in transit over E1 PRI into the legacy PBX, where analog and system phones keep working as before. FreePBX hands call events to the ITGRIX connector over the Asterisk interfaces (AMI/ARI), and the connector works with Bitrix24 over REST API. Only call data goes to the cloud — the voice traffic itself stays on-premise inside your network; there’s no loop through Bitrix24 cloud telephony. Highlighted in blue is what we supply and configure — the gateway and the connector.

Hover or tap a scheme element — we’ll show what it does

  1. Alvis is inserted in-line on the city trunk

    The E1 trunk from the carrier enters not the PBX but the gateway — it sits between the city and the legacy PBX. The gateway then decides on its own where to send a call: to FreePBX over SIP, or in transit to the PBX over E1. It speaks the PBX’s language — EDSS1, QSIG, SS7 — with hardware echo cancellation and route redundancy.

  2. The legacy PBX becomes a large, reliable analog gateway

    From the core of your telephony, the PBX turns into what it’s still good at: holding analog lines, executives' system handsets, the door intercom, and the fax. Nobody re-wires the cross-connect and no handsets are replaced. The call logic is taken off it — and with it go the vendor’s per-port and per-feature licenses.

  3. FreePBX becomes the new call logic

    This is where IVR, queues, schedules, call recording, conferencing, and softphones on laptop and mobile live. The key part — mixed subscriber groups: a single queue or pickup group happily combines an IP phone in the office, an analog handset behind the legacy PBX, and an employee’s mobile out in the field. FreePBX sees everyone and manages everyone — no matter what hardware a person is connected to. There are no per-extension or per-feature licenses.

  4. The ITGRIX connector links Asterisk with Bitrix24 — without a voice loop to the cloud

    The module listens to call events over the Asterisk interfaces and works with the portal via REST API. An inbound call pops the customer card before the handset is picked up, a missed call turns into a lead with an assignee, the recording lands in the deal’s timeline, and calling from the card takes one click.

    The voice never goes to the cloud. Only call data is passed to Bitrix24 — number, direction, duration, link to the recording. The media stream stays inside your network instead of being shuttled to the cloud and back: no extra latency, no dependence of call quality on your internet link, and no charge for the portal’s cloud telephony. Works with both cloud Bitrix24 and the on-premise edition.

Who gets what

What everyone involved in the decision gets

Three people sign off on the project, and each has their own criterion. Below is exactly what changes for each of them.

CEO Cost and control
  • Savings on replacing the switch. A new PBX of the same capacity costs millions. The existing switch is already paid for, and there is no need to scrap it: you pay for one gateway, one server, and the installation work.
  • A smooth migration, no big-bang cutover. No day-one moment when the whole company wakes up to a new phone system. Both systems work as one: the familiar numbering is preserved, employees move to IP phones department by department, and slots on the old switch are freed up gradually. You can pause and consolidate at any step - no one is ever left without service.
  • Savings on the calls themselves. Long-distance and international traffic goes over cost-effective SIP routes. Expensive local E1 trunks and leased inter-office links are replaced with SIP - and nothing has to be reconfigured on the old PBX side.
  • Adding capacity stops being a project. A new employee means an extension in 10 minutes, not ordering a board or an expansion shelf from the vendor - which often can no longer be bought at all.
  • Visibility into what happens with inbound calls. How many calls, how many were lost, which hours are understaffed - a report, not guesswork.
  • Less dependence on a single person. FreePBX and Bitrix24 can be serviced by any qualified contractor, not the one engineer who happens to remember your switch.
IT Director Operations and risk
  • Reversibility. The old PBX physically stays in the loop. If something goes wrong, the route falls back to it, rather than a week-long rollback.
  • E1 transit does not degrade. The gateway has a hardware timeslot cross-connect: transit E1↔E1 calls bypass the processor. No added latency and no loss of quality on the segment you did not touch.
  • Route redundancy in the gateway itself. If the SIP leg becomes unavailable, Alvis reroutes calls over a backup route with no action on your part.
  • No free E1 board - not a problem. A PRI board can be added; if all slots are taken, one FXS slot is freed and its subscribers move to IP phones. For the common Panasonic and LG models, boards are supplied turnkey.
  • Equipment from the manufacturer, not a reseller. The gateways are ours: always in stock, delivered within a week, with a replacement pool and incident analysis handled by the very people who designed them. No distributor-to-overseas-vendor chain.
  • Manageable operations. Linux, a web interface, configs under version control, backups of settings and recordings with built-in tools.
  • No vendor lock-in. Open Asterisk/FreePBX, standard SIP, a documented Bitrix24 REST API. You can switch contractors.
Head of Sales Inbound calls and quality
  • A missed call = a lead. A customer who could not get through automatically becomes a task with an assigned manager and a deadline, instead of disappearing into the switch log.
  • The card before hello. The manager sees the company, the deal, and past agreements while the call is still ringing.
  • Review from the recording. A disputed conversation, onboarding a new hire, checking a script - the recording is attached to the deal, nothing to search for.
  • Distribution rules. A returning customer reaches their own manager, a new one goes into a load-based queue, and a night call goes to voicemail with a task for the morning.
  • The full picture across all employees at once. The CRM captures calls both from those who have already moved to IP phones and from those still on an analog handset behind the old switch. There is no need to wait until everyone is migrated.

Under the hood

Technical capabilities

A section for the IT team: exactly what is supported and to what extent.

E1 interface - the Alvis gateway

  • In-line on the E1 trunk: the local trunk enters the gateway rather than going straight to the switch. Some calls go to FreePBX over SIP, the rest transit over E1 to the old PBX; the rules are set on the gateway itself. Toward the public network the gateway acts as user, toward the PBX as net.
  • Subscriber names and QSIG: QSIG mode is enabled on the trunk to the PBX, so not only numbers but subscriber names reach the switch. For a specific PBX model - Siemens, Avaya - the signaling is tailored individually.
  • Hardware timeslot cross-connect: E1↔E1 transit is switched in hardware, bypassing the processor - audio quality does not degrade and no extra latency appears on the link.
  • Hardware echo cancellation: handled on the DSP rather than loading the FreePBX server.
  • A proven hardware platform: the original hardware design is licensed from Odin TeleSystems Inc. - hence the stability and reliability of the E1 interface. Standalone, fanless operation with low power consumption.

Connector ITGRIX for Bitrix24

  • A separate module between Asterisk and the portal: it picks up call events through Asterisk’s interfaces (AMI/ARI) and works with the CRM over the REST API.
  • No audio loop into the cloud: only call data goes to Bitrix24, while the media stream stays inside your network. No added latency, no dependence of call quality on your internet link, and no charge for the portal’s cloud telephony.
  • Data on all subscribers, not just the new ones: the CRM captures calls both from those already on IP phones and from those still on analog handsets behind the old PBX. There is no need to migrate everyone at once to see the whole picture.
  • A screen pop with the customer, the deal, and the assigned manager before you pick up.
  • A lead from a missed call with a task created automatically and a response deadline.
  • Click-to-call: call straight from the card, no need to dial the number.
  • A recording in the deal’s timeline - listen from within the CRM, with no access to the telephony server.
  • Routing by assigned manager: a customer’s repeat call goes to their manager, or to the queue if that manager is unavailable.
  • Compatible with cloud Bitrix24 and with the on-premise edition on your own server.

FreePBX - call logic

  • Routing: IVR of any depth, schedules and holidays, inbound routes by number and by time.
  • Mixed subscriber groups: an IP phone, an analog handset behind the old PBX, and an employee’s mobile can all work in the same queue or pickup group. FreePBX manages them all regardless of which hardware they are connected to.
  • Queues instead of ring groups: a call does not ring every phone at once - it enters a queue and is distributed by strategy: evenly, by load, or by priority. The caller does not hear a busy tone: they are told their position and the expected wait time.
  • A pause between calls (wrap-up): a configurable interval, usually 5 seconds, so the operator can finish the card and catch their breath instead of taking the next call mid-sentence.
  • Call recording: selective or complete, stored on your own server, access governed by permissions.
  • FMC for field employees: a corporate SIM card becomes an extension and works over the carrier’s voice network rather than the internet - there is connectivity where a softphone is useless: weak signal, exhausted data, mobile data switched off. The original CallerID is passed through: the employee sees the customer’s real number, while the corporate number - not a personal mobile - is shown to the outside.
  • A unified numbering plan: a shared 3- or 4-digit numbering scheme for the head office and branches. Familiar numbers on the old switch are still preserved - no need to retrain staff.

Server side and operations

  • Deployment: a physical server or a virtual machine within your environment - ESXi, Hyper-V, and other hypervisors.
  • Resources, as a guide: up to 100 subscribers - 2 vCPU / 4 GB / 60 GB; up to 300 - 4 vCPU / 8 GB / 250 GB plus a disk for recordings.
  • Your data stays with you: call recordings and CDRs are kept on your own hardware - important for data-protection requirements.
  • Network: a dedicated VLAN for voice, QoS/DSCP on the switches, static addresses for the gateway and the server.
  • Fault tolerance: a second node with configuration replication and failover; plus a backup route on the gateway itself.
  • Backup: a daily backup of the configuration and database, with a separate retention policy for recordings.

Connector

Benefits of the ITGRIX connector

The ITGRIX connector links Asterisk with Bitrix24 directly, with no audio loop through the cloud. Below are real screens from a live portal: exactly what shows up in the CRM.

ITGRIX connector: automatic lead creation from an incoming call in Bitrix24 with call recording

Automatic lead creation

An unknown number calls in and the connector creates a lead on its own, with source and assigned manager. Not a single call is lost, even if the manager couldn’t pick up.

Learn more at itgrix.ru →
ITGRIX connector: auto-routing an incoming call to the assigned manager in Bitrix24

Auto-routing to the assigned manager

A returning customer’s call goes straight to their manager instead of the general queue, so the conversation continues with someone who already knows the history.

Learn more at itgrix.ru →
ITGRIX connector: a call event in a Bitrix24 card with duration, wait time and recording

Logging calls in Bitrix24

Every call is saved in the customer card: duration, time waited for an answer, and a call recording you can play back right from the CRM.

Learn more at itgrix.ru →
ITGRIX connector: a missed call turns into a lead and a call-back task in Bitrix24

Capturing missed calls

A missed call turns into a lead and a “call back” task with an assigned manager and a response deadline, so the customer is neither forgotten nor lost.

Learn more at itgrix.ru →
ITGRIX connector: click-to-call a number from the customer card through the softphone

Click-to-call from the CRM

Click a number in the card and the call goes out through the softphone. No manual dialing, and no chance of misdialing.

Learn more at itgrix.ru →
ITGRIX connector: a screen pop with the customer card on a call — activities, deals, assigned manager, quick actions

Screen pop on a call

Before picking up, the manager sees the whole customer: activities, deals, assigned manager, and quick actions — comment, deal, invoice.

Learn more at itgrix.ru →
ITGRIX connector: end-to-end analytics and call tracking — sources, channel performance, conversion

End-to-end analytics

Call tracking: numbers mapped to advertising sources, channel performance, and conversion of calls into sales.

Learn more at itgrix.ru →
ITGRIX connector: module setup and customization — logging, fields, MP3, black/white lists

Extensive customization options

Flexible module configuration: routing rules, field handling, conversion of recordings to MP3, black/white lists, and admin commands.

Learn more at itgrix.ru →
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Roadmap

Implementation stages

About 1–3 weeks in calendar time overall. We are the manufacturer, and the equipment sits in our own warehouse — there’s no waiting months for delivery, and the on-site work takes just a few days. Some stages run in parallel: while the specification is being agreed, the equipment is already being assembled.

  1. 1–2 days

    Audit

    We look at the PBX make, free E1 ports, signaling type, cross-connect, power, and network. We count the real number of extensions and simultaneous calls. The result is a report: what can be kept and what has to be replaced.

  2. 1–2 days

    Design and cost estimate

    Connection diagram, numbering plan, routing scenarios, equipment list, an exact cost estimate, and a schedule. We align it with your IT team before purchasing, not after.

  3. up to 5 days

    Delivery

    An Alvis gateway and, if needed, SIP phones and headsets. We manufacture the gateways ourselves and keep them in stock — no waiting for them from abroad or ordering them per project. Procurement and shipping run in parallel with approving the design, not after it. A server usually isn’t purchased: we deploy FreePBX on your existing hardware.

  4. 2–4 days

    Commissioning

    We set up FreePBX, bring up the E1 link on the gateway, and configure routes, IVR, queues, and recording. We test it on a staging setup while your live phone service runs as usual.

  5. 1 day

    Cutover · Bitrix24

    The switchover happens in an agreed cutover window — usually 30–60 minutes. We connect Bitrix24 and verify the customer card, a lead from a missed call, click-to-call, and recordings in the deal timeline. We train the administrator and the users.

  6. at your own pace

    Migration and support

    From there, migration proceeds without rush and without large one-off costs: departments move to hardware IP phones one at a time, slots on the old PBX are gradually freed up, and the familiar numbering is preserved. You can stop and settle in at any step. In parallel: monitoring, updates, changes to scenarios, and adding numbers; the response SLA is fixed in the contract.

From our practice

Typical implementation scenarios

Three configurations that come up most often. Find yours — the scope of work and the ballpark figures will be close.

The scenarios are generalized: these are typical project configurations, not case studies of specific customers. Real references with contacts are provided on request during negotiations.

Estimate it yourself

Calculator: budget, timeline and what to prepare

Four parameters — and you get ballpark figures, a realistic timeline, and a list of what to check on your side before work begins.

How many employees should get an internal number after the upgrade.

How many E1 trunks to bring into the new PBX. One trunk — up to 30 simultaneous calls.

Basic covers 80% of a sales team’s needs. Advanced is needed when you have your own call-routing rules.

Additional options

Estimated budget

—

one-time costs: hardware, work and integration. Extensions and call recording don’t affect the total — there are no per-extension or per-feature licenses in FreePBX.

Donut chart: share of estimate line items

What the total is made of. Hover over an estimate line — the segment highlights.

Estimate breakdown

Deployment timeline

—

from audit to traffic cutover

Checklist: what to prepare

    This estimate is approximate and is not an offer. An exact quote follows the audit: the price depends on the PBX brand, signaling type, cross-connect condition, and recording storage requirements.

    Send the estimate and get an exact quote

    How much it costs

    Implementation options

    Three standard configurations. The amounts are one-time: equipment, labor, and integration. There are no per-extension licenses or subscription fees for FreePBX features. “From” is the lower bound of the same estimate as in the calculator above.

    Basic

    Replace the core and remove dependence on the old system

    from $2,000

    Configuration: 2×E1, FreePBX on your infrastructure, no CRM integration. Extensions — as many as needed

    • Alvis-GW-2E1 gateway, up to 60 concurrent calls
    • FreePBX on your existing server or in your virtualization
    • IVR, queues, groups, schedules
    • Softphone and remote employees
    • The old PBX stays as a backup
    • Call recording — built-in, no licenses
    • Bitrix24 integration
    • High availability
    Discuss

    Most popular

    Standard

    Full-featured telephony with CRM and recording

    from $2,700

    Configuration: 2×E1, FreePBX on your infrastructure, basic integration, call recording

    • Everything in Basic
    • Bitrix24 integration: contact card, click-to-call
    • Lead from a missed call with an assigned owner
    • Call recordings in the deal timeline
    • Reports on calls and queues
    • Training for the administrator and users
    • Advanced distribution scenarios
    • High availability
    Discuss

    High-availability

    When communication downtime is unacceptable

    from $7,000

    Configuration: 4×E1, two dedicated servers, advanced integration, recording

    • Everything in Standard
    • Two dedicated servers and automatic failover
    • Backup routes on the gateway
    • Advanced routing scenarios
    • Distribution by Bitrix24 owners
    • Response times defined in the contract (SLA)
    • Monitoring and daily backup
    • Priority support
    Discuss
    What each option includes
    Feature Basic Standard High-availability
    Keeping your numbers and E1 streamyesyesyes
    Existing phones, no replacementyesyesyes
    Extensionsas many as neededas many as neededas many as needed
    Per-number and per-feature licensesnonono
    E1 streams224
    FreePBX hostingyour infrastructureyour infrastructuretwo dedicated servers
    IVR, queues, schedulesyesyesyes
    Softphone, FMC, remote employeesyesyesyes
    Call recordingyesyesyes
    Bitrix24 integrationnobasicadvanced
    Lead from a missed callnoyesyes
    Second node and automatic failovernonoyes
    Response SLAon requeston requestin the contract
    Implementation time1–3 weeks1–3 weeks3–5 weeks

    Prices are approximate, apply to standard configurations, and do not constitute a public offer. Post-delivery support is priced separately and depends on the required response time.

    First step

    Let’s start with an audit

    Send us your PBX brand and approximate number of employees — we’ll tell you whether this setup fits, and give you a ballpark cost. It’s free and carries no obligation.

    • We’ll reply within one business day — with specifics, not “let’s hop on a call.”
    • We’ll send a design for your PBX — what gets connected and what stays as is.
    • We’ll give you a budget range and timeline before any on-site visit.
    • You can take the equipment for a trial — up to a month, no deposit. If it doesn’t fit, send it back.
    • Email alvis@otx.ru
    • Working hours Mon–Fri, 10:00–19:00 MSK
    • Office 150057, Yaroslavl, Slepneva St. 37, office 12

    Audit request

    Fields marked with an asterisk are required. We don’t share your contacts with third parties.

    Straight answers

    Frequently asked questions

    The key questions we hear at the first meeting. Don’t see yours? Use the search or ask us through the form.

    Keeping your numbers

    Yes. We don’t change your carrier and we don’t port numbers — the E1 trunk stays the same, and so does your service contract. The only thing that changes is what happens to a call once it arrives: previously it went to the legacy station, now it reaches FreePBX through the Alvis gateway.

    Replacing equipment

    There’s no need to replace phones. Analog and system phones stay on the legacy PBX and work exactly as before. New IP phones or a softphone are only needed for the employees you move to FreePBX — usually sales, support, and remote workers.

    If you later decide to move everyone, it’s done gradually, phone by phone as needed, rather than all in a single day.

    Interruptions to phone service

    The main work is done on a separate test bench while your production telephony keeps running as usual. The actual cutover is a single agreed window, typically 30–60 minutes, most often early in the morning or after the end of the workday.

    If something goes wrong, the route falls back to the legacy station — it physically stays in place and is not dismantled.

    Phased rollout

    Yes, and it’s a common scenario. The first phase is the gateway, FreePBX, and one department — sales, for example. Everyone else keeps working on the legacy PBX. Once you’re satisfied that everything works, you connect the next department. The Bitrix24 integration can also be deferred to a second phase.

    Carrier compatibility (SIP/E1)

    If your carrier already offers SIP instead of E1, a gateway is still needed, but on the other side. Alvis takes the place of the E1 link being removed: the carrier delivers SIP, and the gateway converts it into E1 for your legacy station so its phones keep working. Nothing has to be reconfigured on the PBX side — it keeps receiving its E1 trunk just as before.

    This is a separate and increasingly common task: providers are migrating customers from trunks to SIP, and the monthly fee for E1 is noticeably higher. As a bonus, you can route long-distance traffic through a more cost-effective SIP operator.

    Storing call recordings

    On your own server, inside your own perimeter. Neither the recordings nor the CDRs leave for a third-party cloud. This matters if you process personal data and are required to comply with Federal Law 152-FZ.

    The retention period, the list of people with access, and the deletion policy are configured according to your internal rules. Playback from Bitrix24 still works — the CRM accesses the recording by link rather than storing it itself.

    Operation during an internet outage

    Internal calls and landline calls over E1 will keep working: both the gateway and FreePBX sit in your own network and don’t need the internet for this. Without internet, only external services stop working — the cloud version of Bitrix24 and remote employees outside the office.

    Post-project support

    Your administrator receives access and training — adding numbers, editing schedules, and IVR changes are done through the web interface without a contractor. Beyond that, under the contract: monitoring, updates, call-flow changes, and incident response times. You choose the scope of support.

    The solution is open: Asterisk/FreePBX, standard SIP, and a documented Bitrix24 REST API. Any qualified contractor can maintain it — no one is locking you in.