GSM to SLN Converter

Re-encode GSM speech into Asterisk SLN linear audio online

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Asterisk PBX Native

Convert GSM telephony speech directly to SLN — the signed linear format used natively by Asterisk for prompts, IVR, and hold music.

Telephony Parameters

Configure sample rate and encoding depth to match your Asterisk PBX version and telephony system requirements.

Secure Processing

GSM uploads are erased immediately. SLN results are purged from our servers within 24 hours.

How to convert GSM to SLN

1

Select files from Computer, Google Drive, Dropbox, URL or by dragging it on the page.

2

Choose sln or any other format you need as a result (more than 200 formats supported)

3

Let the file convert and you can download your sln file right afterwards

About formats

GSM 06.10 (Full Rate) is the foundational speech codec of the Global System for Mobile Communications standard, ratified by ETSI in 1991 and deployed across hundreds of cellular networks worldwide. Operating at a fixed 13 kbit/s, the algorithm applies Regular Pulse Excitation with Long-Term Prediction (RPE-LTP) to compress 20 ms frames of 8 kHz mono speech into just 33 bytes each. This approach models the vocal tract as a linear predictive filter, encodes the excitation signal, and leverages pitch periodicity for further reduction — tuned to deliver intelligible voice under the bandwidth constraints of early digital mobile channels. The codec powers not only GSM telephony but also many VoIP applications, voicemail systems, and IVR platforms that benefit from its low bitrate. Three concrete advantages stand out. First, extraordinary compression: one minute of speech fits in roughly 100 KB, enabling efficient storage and transmission. Second, universal tooling — libraries such as libgsm and SoX handle encoding and decoding on every major platform. Third, a royalty-free patent landscape that has encouraged adoption across open-source telephony projects like Asterisk and FreeSWITCH.
Initial release: 1991
SLN (Signed Linear) is a headerless raw audio format storing 16-bit signed linear PCM samples at 8000 Hz mono, most closely associated with Asterisk) — the open-source PBX framework developed by Digium (now Sangoma Technologies). Within Asterisk, SLN serves as the native internal audio representation: every codec transcoding operation passes through signed linear as an intermediate step. This makes SLN the backbone of Asterisk's codec translation architecture. The format contains nothing but raw samples — no headers, no metadata, no framing — so parameters must be known in advance. While this lack of self-description might seem limiting, it is actually an advantage in telephony where sample format is fixed by convention and every overhead byte matters across thousands of simultaneous channels. The 8000 Hz rate aligns with the G.711 standard for traditional telephony, capturing the full 300-3400 Hz voice band. Asterisk also supports extended variants (sln16, sln32, sln48) for wideband audio. SLN files require no decoding — just direct memory mapping — making them ideal for real-time mixing, conferencing, and prompt playback in high-density VoIP environments.
Initial release: 1999

Frequently Asked Questions

What is SLN?

SLN is Asterisk PBX's signed linear PCM format — used for voice prompts, hold music, and audio playback on Asterisk telephony systems.

Why convert GSM to SLN?

Asterisk PBX uses SLN as its native internal format. Converting GSM to SLN gives you audio ready for Asterisk voice prompts and IVR menus.

What sample rate does SLN use?

Standard Asterisk SLN uses 8000 Hz mono, matching telephony standards. Higher rates are supported in newer Asterisk versions.

Can I play SLN outside Asterisk?

SoX and Audacity can import SLN as raw signed linear PCM by specifying the correct sample rate and encoding manually.

Is my data protected?

GSM uploads are deleted after conversion. SLN outputs are removed from our servers within 24 hours.

GSM to SLN Quality Rating

5.0 (2 votes)
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