| Topic | Replies | Views | Activity | |
|---|---|---|---|---|
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Asterisk Release 23.2.1
The Asterisk Development Team would like to announce the release of asterisk-23.2.1. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The official Asterisk P… |
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1 | 21 | January 29, 2026 |
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Asterisk Release 22.8.1
The Asterisk Development Team would like to announce the release of asterisk-22.8.1. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The official Asterisk P… |
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2 | 20 | January 29, 2026 |
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Asterisk Release 20.18.1
The Asterisk Development Team would like to announce the release of asterisk-20.18.1. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The official Asterisk … |
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2 | 11 | January 29, 2026 |
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Certified Asterisk Release certified-20.7-cert8
The Asterisk Development Team would like to announce the release of Certified asterisk-20.7-cert8. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The offic… |
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2 | 30 | January 12, 2026 |
| Can't hangup an connected call |
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0 | 7 | February 1, 2026 |
| Favorite softphone? |
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3 | 48 | January 31, 2026 |
| Regarding to send the streamId as the headers while making the external Media |
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3 | 23 | January 30, 2026 |
| Webrtc Call issues in Jio network |
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1 | 18 | January 30, 2026 |
| Asterisk - Source not in ICE active candidate list |
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5 | 28 | January 30, 2026 |
| Problem with Asterisk AMI after switching from chan_sip (Asterisk 18.8.0) to pjsip (Asterisk 22.7.0) |
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5 | 48 | January 30, 2026 |
| Windows-Based Android SIP GSM Gateway |
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2 | 37 | January 30, 2026 |
| Request for Guidance on Connecting Cisco 7821 Phones to Asterisk / FreePBX |
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8 | 55 | January 30, 2026 |
| Asterisk 22.2.0: first 5 seconds of call = wavy/cracking audio (jssip + Tele2, works in 3CX) |
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3 | 38 | January 29, 2026 |
| Chrome WebRTC media becomes permanently blocked after first call (even when microphone is allowed) |
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1 | 41 | January 29, 2026 |
| Preserve and forward the original hangup cause to the SIP endpoint |
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3 | 41 | January 28, 2026 |
| How to set a Caller ID from matrix or group |
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2 | 23 | January 27, 2026 |
| Feature Request: Dynamic WebSocket URI Support for ARI ExternalMedia |
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3 | 26 | January 27, 2026 |
| Dropped calls with same number of bots (~25) on weaker and more powerful hardware. |
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5 | 48 | January 26, 2026 |
| Alembic CDR PGSQL Definition NOT compatible with cdr_pgsql.c |
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3 | 40 | January 26, 2026 |
| Using Asterisk with ICTContact for an Omni-Channel Contact Center |
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2 | 20 | January 27, 2026 |
| Problem 22.6 --> 22.8 pjsip |
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1 | 68 | January 25, 2026 |
| VEXYL - Self-Hosted AI Voice Gateway for Asterisk |
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10 | 121 | January 25, 2026 |
| Asterisk as a backup telephony system |
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3 | 41 | January 23, 2026 |
| Asterisk Release 23.2.0 |
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3 | 21 | January 29, 2026 |
| Asterisk Release 22.8.0 |
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3 | 26 | January 29, 2026 |
| Asterisk Release 20.18.0 |
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3 | 30 | January 29, 2026 |
| Multi-Tenant Contact Center Challenges We Faced While Building ICTContact on Asterisk |
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2 | 26 | January 27, 2026 |
| How to maintain TRANSFER_CONTEXT between Attended Transfers? |
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3 | 44 | January 21, 2026 |
| Unable to change the control message format |
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5 | 17 | January 21, 2026 |
| What problems do you face with multi-tenant contact center software? |
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0 | 20 | January 21, 2026 |