E1 for Asterisk and FreePBX

The Alvis gateway processes E1 PRI, converts it to SIP and connects to Asterisk as a SIP trunk. Thanks to the built-in statistics module, Alvis logs all Layer 1 problems, which helps with connection. The built-in frequency meter finds typical operator problems when TDMoE is used instead of "honest E1" — carrier drift, carrier loss and jitter, while the line echo canceller removes local and network echo up to 128 ms.

If Alvis and Asterisk are on the same network, a trunk without registration is recommended — it simplifies configuration on both sides. On the Asterisk side, remember to set insecure=port,invite and directmedia=no on the trunk.

A single E1 stream. There is a corporate Asterisk with several provider SIP trunks, and you also need to feed in E1. It makes sense to use the Alvis-GW-2E1-Lite: all full-version functions except E1-E1 transit connections (not needed here), plus discounts for Asterisk users. Even in a 1×E1 configuration it comes out cheaper than the nearest competitors thanks to an optimal hardware configuration, refined circuitry and proven software.

Two E1 streams from different providers. There is an Asterisk-based call center; you need to feed 2 E1 streams with maximum reliability and redundancy. We install a rack Alvis-GW-2x2E1-R2-Lite with two independent GW2 modules: each stream goes to its own processor module separately, so there is no E1-synchronization problem from different sources. In competing devices, a large carrier-frequency difference causes SLIP, Loss Frame Alignment and PRI D-channel drops up to call breakup and "freezing" — here the streams are separate and there is no problem. This opens up building redundant FailOver E1/SIP systems with the Rescue-4E1 device. Even if one processor module fails, the system switches E1 to the other — calls are not lost. Asterisk users also get discounts on this model.

There is a Panasonic TDA100, LG LDK300, Siemens HiPath3000 or similar system that has worked for years. Management decided to put Asterisk at the center: they need call recording, a modern IVR, IP phones (including for remote employees and entire offices) — but they want to keep the existing PBX.

Solution: we connect an Alvis-GW-2E1 (-D, -L, -R1). The existing PBX is left purely as a large FXS gateway, all logic moves to Asterisk, and we disable DISA and other proprietary features of the old PBX. Alvis connects over PRI in QSIG mode — this correctly passes not only numbers but also subscriber names. Alvis processes E1 PRI from the PBX, converts it to SIP and connects to Asterisk as a SIP trunk.

Advantages: a single numbering plan is configured (3 or 4 digits), subscribers of both systems call each other; SIP-operator trunks are set up on Asterisk — the company stops overpaying for long distance, 8-800 numbers are connected, CRM is integrated. GarantPlus specialists help connect to the PBX, including on the Panasonic/LG side and other common types. More about PBX modernization in the dedicated article.

Three-way connectivity is configured between the PSTN (E1), PBX (E1) and SIP networks: long-distance/intercity PSTN over SIP plus SIP subscribers. The most important thing here is a well-thought-out dial plan and routing; thanks to Alvis’s well-designed routing system this task is greatly simplified.