# Latest **URL:** https://community.asterisk.org/latest.md [Latest](https://community.asterisk.org/latest.md) · [Categories](https://community.asterisk.org/categories.md) --- ## [AstriCon 2027 🧑‍🚀 Call For Speakers](https://community.asterisk.org/t/astricon-2027-call-for-speakers/114628)
**Author:** [@penguinpbx](https://community.asterisk.org/u/penguinpbx)\ **Replies:** 0\ **Last updated:** [October 6, 2026, 3:38pm UTC](https://community.asterisk.org/t/astricon-2027-call-for-speakers/114628 "2026-10-06T15:38:07Z")
AstriCon returns to Pasadena this spring as part of SCaLE - deadline to submit your speaking proposal is 2026-12-31T23:59:00Z (UTC) - more details in this Asterisk blog post by YT :cowboy\_hat\_face: --- ## [Asterisk Release 22.11.0](https://community.asterisk.org/t/asterisk-release-22-11-0/114073)
**Author:** [@asteriskteam](https://community.asterisk.org/u/asteriskteam)\ **Replies:** 0\ **Last updated:** [August 27, 2026, 3:30pm UTC](https://community.asterisk.org/t/asterisk-release-22-11-0/114073 "2026-08-27T15:30:22Z")
The Asterisk Development Team would like to announce the release of asterisk-22.11.0. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The official Asterisk … --- ## [Asterisk Release 23.5.0](https://community.asterisk.org/t/asterisk-release-23-5-0/114072)
**Author:** [@asteriskteam](https://community.asterisk.org/u/asteriskteam)\ **Replies:** 0\ **Last updated:** [August 27, 2026, 3:29pm UTC](https://community.asterisk.org/t/asterisk-release-23-5-0/114072 "2026-08-27T15:29:53Z")
The Asterisk Development Team would like to announce the release of asterisk-23.5.0. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The official Asterisk P… --- ## [Asterisk Release 20.21.0](https://community.asterisk.org/t/asterisk-release-20-21-0/114071)
**Author:** [@asteriskteam](https://community.asterisk.org/u/asteriskteam)\ **Replies:** 0\ **Last updated:** [August 27, 2026, 3:29pm UTC](https://community.asterisk.org/t/asterisk-release-20-21-0/114071 "2026-08-27T15:29:20Z")
The Asterisk Development Team would like to announce the release of asterisk-20.21.0. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The official Asterisk … --- ## [Certified Asterisk Release certified-22.8-cert4](https://community.asterisk.org/t/certified-asterisk-release-certified-22-8-cert4/113834)
**Author:** [@asteriskteam](https://community.asterisk.org/u/asteriskteam)\ **Replies:** 0\ **Last updated:** [August 7, 2026, 3:47pm UTC](https://community.asterisk.org/t/certified-asterisk-release-certified-22-8-cert4/113834 "2026-08-07T15:47:31Z")
The Asterisk Development Team would like to announce the release of Certified asterisk-22.8-cert4. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The offic… --- ## [Multi-tenant hotel and restaurant PBX on Asterisk with PJSIP realtime (Filantis): design notes and three fixes](https://community.asterisk.org/t/multi-tenant-hotel-and-restaurant-pbx-on-asterisk-with-pjsip-realtime-filantis-design-notes-and-three-fixes/114752)
**Author:** [@keshavdutt](https://community.asterisk.org/u/keshavdutt)\ **Replies:** 2\ **Last updated:** [October 13, 2026, 9:37pm UTC](https://community.asterisk.org/t/multi-tenant-hotel-and-restaurant-pbx-on-asterisk-with-pjsip-realtime-filantis-design-notes-and-three-fixes/114752 "2026-10-13T21:37:17Z")
Disclosure: I’m the developer of Filantis, a commercial product, which is why this is in the Commercial category. Filantis is a multi-tenant business phone system (cloud PBX or on-premise) built on Asterisk, with hotel,… --- ## [SIP configuration problem](https://community.asterisk.org/t/sip-configuration-problem/114755)
**Author:** [@mado\_0699](https://community.asterisk.org/u/mado_0699)\ **Replies:** 2\ **Last updated:** [October 13, 2026, 4:12pm UTC](https://community.asterisk.org/t/sip-configuration-problem/114755 "2026-10-13T16:12:09Z")
Hello, I'm currently having a problem with a SIP extension. Context: Two servers are being used as SIP extensions; one works and the other doesn't. (Same VLAN and same network settings) During testing, the connection is… --- ## [looking for advice on multi tenant serviced office ](https://community.asterisk.org/t/looking-for-advice-on-multi-tenant-serviced-office/114670)
**Author:** [@Foneextra](https://community.asterisk.org/u/Foneextra)\ **Replies:** 6\ **Last updated:** [October 13, 2026, 3:38pm UTC](https://community.asterisk.org/t/looking-for-advice-on-multi-tenant-serviced-office/114670 "2026-10-13T15:38:48Z")
Not sure if this is the right place to put this I’m looking for some advice on the best way to build a modern multi-tenant VoIP system using Asterisk for a serviced-office environment. We currently operate serviced off… --- ## [Vicidial collaboration](https://community.asterisk.org/t/vicidial-collaboration/114709)
**Author:** [@suzairabbas](https://community.asterisk.org/u/suzairabbas)\ **Replies:** 1\ **Last updated:** [October 12, 2026, 10:43pm UTC](https://community.asterisk.org/t/vicidial-collaboration/114709 "2026-10-12T22:43:27Z")
:rocket: VICIDIAL AUTO DIALER SERVICES | SETUP, CONFIGURATION & COLLABORATION :rocket: Looking for a reliable VICIdial Auto Dialer solution for your call center or business? I can help you get started! :laptop:… --- ## [Does ARI event streaming with Stasis not work in prod and am i compelled to use Chan\_websocket?](https://community.asterisk.org/t/does-ari-event-streaming-with-stasis-not-work-in-prod-and-am-i-compelled-to-use-chan-websocket/114668)
**Author:** [@jatin](https://community.asterisk.org/u/jatin)\ **Replies:** 2\ **Last updated:** [October 8, 2026, 1:23pm UTC](https://community.asterisk.org/t/does-ari-event-streaming-with-stasis-not-work-in-prod-and-am-i-compelled-to-use-chan-websocket/114668 "2026-10-08T13:23:21Z")
my bad i am a fresher and never worked in prod especially with asterisk i have been doing R&D with asterisk and working with ari with event listening where i moved the whole control of my call to the python application w… --- ## [Asterisk Release Candidate 24.0.0-rc2](https://community.asterisk.org/t/asterisk-release-candidate-24-0-0-rc2/114676)
**Author:** [@asteriskteam](https://community.asterisk.org/u/asteriskteam)\ **Replies:** 0\ **Last updated:** [October 8, 2026, 12:20pm UTC](https://community.asterisk.org/t/asterisk-release-candidate-24-0-0-rc2/114676 "2026-10-08T12:20:34Z")
The Asterisk Development Team would like to announce release candidate 2 of asterisk-24.0.0. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The official As… --- ## [Asterisk Release Candidate 23.6.0-rc2](https://community.asterisk.org/t/asterisk-release-candidate-23-6-0-rc2/114675)
**Author:** [@asteriskteam](https://community.asterisk.org/u/asteriskteam)\ **Replies:** 0\ **Last updated:** [October 8, 2026, 12:18pm UTC](https://community.asterisk.org/t/asterisk-release-candidate-23-6-0-rc2/114675 "2026-10-08T12:18:55Z")
The Asterisk Development Team would like to announce release candidate 2 of asterisk-23.6.0. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The official As… --- ## [Asterisk Release Candidate 22.12.0-rc2](https://community.asterisk.org/t/asterisk-release-candidate-22-12-0-rc2/114674)
**Author:** [@asteriskteam](https://community.asterisk.org/u/asteriskteam)\ **Replies:** 0\ **Last updated:** [October 8, 2026, 12:15pm UTC](https://community.asterisk.org/t/asterisk-release-candidate-22-12-0-rc2/114674 "2026-10-08T12:15:49Z")
The Asterisk Development Team would like to announce release candidate 2 of asterisk-22.12.0. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The official A… --- ## [Asterisk Release Candidate 20.22.0-rc2](https://community.asterisk.org/t/asterisk-release-candidate-20-22-0-rc2/114673)
**Author:** [@asteriskteam](https://community.asterisk.org/u/asteriskteam)\ **Replies:** 0\ **Last updated:** [October 8, 2026, 12:12pm UTC](https://community.asterisk.org/t/asterisk-release-candidate-20-22-0-rc2/114673 "2026-10-08T12:12:20Z")
The Asterisk Development Team would like to announce release candidate 2 of asterisk-20.22.0. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The official A… --- ## [NOTIFY for BLF/Pickup missing Remote-Part](https://community.asterisk.org/t/notify-for-blf-pickup-missing-remote-part/114661)
**Author:** [@kwemheuer](https://community.asterisk.org/u/kwemheuer)\ **Replies:** 4\ **Last updated:** [October 8, 2026, 9:19am UTC](https://community.asterisk.org/t/notify-for-blf-pickup-missing-remote-part/114661 "2026-10-08T09:19:34Z")
Hi, I’m having a problem with notifications for BLF and Pickup. In version 22.6 (and ealier), the SDP body contained a “Remote” section and a ‘Local’ section. In version 22.11, the “Remote” section is missing. A workin… --- ## [Sending SIP Notify to specific PJSIP contact](https://community.asterisk.org/t/sending-sip-notify-to-specific-pjsip-contact/114585)
**Author:** [@oza4h07](https://community.asterisk.org/u/oza4h07)\ **Replies:** 5\ **Last updated:** [October 8, 2026, 9:18am UTC](https://community.asterisk.org/t/sending-sip-notify-to-specific-pjsip-contact/114585 "2026-10-08T09:18:42Z")
Hello, I’m thinking about dedicating a specific endpoint to “un-associated” SIP hardphones. By “un-associated”, I mean phones not yet associated to any useful phone number. This specific endpoint should relate 0, 1 or … --- ## [Proposal: report RTP inactivity to ARI/AMI instead of hanging up (rtp\_timeout)](https://community.asterisk.org/t/proposal-report-rtp-inactivity-to-ari-ami-instead-of-hanging-up-rtp-timeout/114637)
**Author:** [@Orobets](https://community.asterisk.org/u/Orobets)\ **Replies:** 5\ **Last updated:** [October 7, 2026, 12:38pm UTC](https://community.asterisk.org/t/proposal-report-rtp-inactivity-to-ari-ami-instead-of-hanging-up-rtp-timeout/114637 "2026-10-07T12:38:08Z")
Hello, Following a suggestion on GitHub issue #2202 (\[new-feature\]: res\_pjsip\_sdp\_rtp: optionally report RTP inactivity to the channel's application instead of hanging up · Issue #2202 · asterisk/asterisk · GitHub), I … --- ## [PBXonix: looking for five administrators to test PBX and SIP trunk monitoring](https://community.asterisk.org/t/pbxonix-looking-for-five-administrators-to-test-pbx-and-sip-trunk-monitoring/114632)
**Author:** [@mikhilgermov-afk](https://community.asterisk.org/u/mikhilgermov-afk)\ **Replies:** 0\ **Last updated:** [October 7, 2026, 10:10am UTC](https://community.asterisk.org/t/pbxonix-looking-for-five-administrators-to-test-pbx-and-sip-trunk-monitoring/114632 "2026-10-07T10:10:28Z")
I am building PBXonix, a cloud monitoring service for Asterisk, FreePBX and Issabel. I would like to get feedback from five administrators using real PBXs. The starting use case is simple: see an unavailable SIP/PJSIP t… --- ## [I’m unable to use RFC4733](https://community.asterisk.org/t/i-m-unable-to-use-rfc4733/114580)
**Author:** [@kaitokun](https://community.asterisk.org/u/kaitokun)\ **Replies:** 1\ **Last updated:** [October 5, 2026, 8:22am UTC](https://community.asterisk.org/t/i-m-unable-to-use-rfc4733/114580 "2026-10-05T08:22:32Z")
Title: RFC4733 DTMF not detected with PJSIP SIP trunk, but in-band DTMF works Hello everyone, I am having trouble receiving RFC4733 DTMF through a SIP trunk using chan\_pjsip. My setup: Asterisk version: \[Asterisk ve… --- ## [Inbound Registration Issue – No SIP Response](https://community.asterisk.org/t/inbound-registration-issue-no-sip-response/114339)
**Author:** [@Hassan1](https://community.asterisk.org/u/Hassan1)\ **Replies:** 11\ **Last updated:** [October 1, 2026, 8:33am UTC](https://community.asterisk.org/t/inbound-registration-issue-no-sip-response/114339 "2026-10-01T08:33:16Z")
Hello everyone, I’m trying to configure Siperb Inbound Registration with an Issabel PBX running Asterisk 18.19.0 and chan\_sip. The goal is to use Siperb Proxy Registration Mode for mobile push notifications. Siperb co… --- ## [Asterisk 20.20 Crashed](https://community.asterisk.org/t/asterisk-20-20-crashed/114562)
**Author:** [@faqterson](https://community.asterisk.org/u/faqterson)\ **Replies:** 2\ **Last updated:** [October 1, 2026, 8:09am UTC](https://community.asterisk.org/t/asterisk-20-20-crashed/114562 "2026-10-01T08:09:20Z")
Hi I receive the following couple minutes before asterisk stopped responding: \[Oct 1 08:48:19\] ERROR\[1915\] stasis.c: FRACK!, Failed assertion user\_data is NULL (0) \[Oct 1 08:48:19\] ERROR\[1915\] : Got 25 backtrace re… --- ## [Bypassing RTP on an Asterisk routing node before and after IVR-to-Genesys transfer](https://community.asterisk.org/t/bypassing-rtp-on-an-asterisk-routing-node-before-and-after-ivr-to-genesys-transfer/114549)
**Author:** [@mfb14](https://community.asterisk.org/u/mfb14)\ **Replies:** 1\ **Last updated:** [September 30, 2026, 8:31am UTC](https://community.asterisk.org/t/bypassing-rtp-on-an-asterisk-routing-node-before-and-after-ivr-to-genesys-transfer/114549 "2026-09-30T08:31:20Z")
Hi everyone, I have some issues about Asterisk RTP traffic. We have the following call-routing topology: SBC → Asterisk 1 (Work as a loadbalancer, route call to other asterisk) → Asterisk IVR (Our ivr runs via AGI We … --- ## [Asterix for a prop phone system / escape room](https://community.asterisk.org/t/asterix-for-a-prop-phone-system-escape-room/114531)
**Author:** [@PhilippeThunderbolt](https://community.asterisk.org/u/PhilippeThunderbolt)\ **Replies:** 4\ **Last updated:** [September 29, 2026, 5:02pm UTC](https://community.asterisk.org/t/asterix-for-a-prop-phone-system-escape-room/114531 "2026-09-29T17:02:58Z")
I’m building an interactive prop telephone for a phone booth in a British pub. Guests will dial fictional numbers to hear recordings, and staff will trigger incoming calls that play a recording when someone answers. May… --- ## [\[Open Source\] AiPBX: Modern enterprise IP telephony platform powered by Asterisk 22](https://community.asterisk.org/t/open-source-aipbx-modern-enterprise-ip-telephony-platform-powered-by-asterisk-22/114500)
**Author:** [@mahirgul](https://community.asterisk.org/u/mahirgul)\ **Replies:** 0\ **Last updated:** [September 28, 2026, 9:13am UTC](https://community.asterisk.org/t/open-source-aipbx-modern-enterprise-ip-telephony-platform-powered-by-asterisk-22/114500 "2026-09-28T09:13:51Z")
Hi everyone, I would like to share an open-source project I have been working on called AiPBX. AiPBX is an enterprise-grade IP telephony platform designed to modernize PBX management while leveraging Asterisk 22 at its… --- ## [I replaced "press 1 for sales, 2 for support" with Jev](https://community.asterisk.org/t/i-replaced-press-1-for-sales-2-for-support-with-jev/114477)
**Author:** [@rayan](https://community.asterisk.org/u/rayan)\ **Replies:** 0\ **Last updated:** [September 26, 2026, 4:54am UTC](https://community.asterisk.org/t/i-replaced-press-1-for-sales-2-for-support-with-jev/114477 "2026-09-26T04:54:00Z")
Hy everyone, I built a lightweight router using TypeSafe’s Jev model. callers speak naturally (“Hello, I’d like to speak to someone”) Jev classifies intent in ~150ms and return confidence score The PBX routes t… --- ## [MixMonitor recordings shorter than call and skipping words: "Flushing audiohook" / "both fail to provide 160 samples" with ARI ExternalMedia in a mixing bridge (Asterisk 16.8.3)](https://community.asterisk.org/t/mixmonitor-recordings-shorter-than-call-and-skipping-words-flushing-audiohook-both-fail-to-provide-160-samples-with-ari-externalmedia-in-a-mixing-bridge-asterisk-16-8-3/114466)
**Author:** [@gauravs456](https://community.asterisk.org/u/gauravs456)\ **Replies:** 0\ **Last updated:** [September 25, 2026, 2:08pm UTC](https://community.asterisk.org/t/mixmonitor-recordings-shorter-than-call-and-skipping-words-flushing-audiohook-both-fail-to-provide-160-samples-with-ari-externalmedia-in-a-mixing-bridge-asterisk-16-8-3/114466 "2026-09-25T14:08:44Z")
Environment Asterisk 16.8.3, chan\_pjsip, ARI Caller: carrier → OpenSIPS → Asterisk (PJSIP), ulaw, no transcoding anywhere Voice bot connected through ARI ExternalMedia (UnicastRTP, ulaw, RTP/UDP, connection\_type: clien… --- ## [Wrote an Asterisk GUI that automatically bills with FreeAgent](https://community.asterisk.org/t/wrote-an-asterisk-gui-that-automatically-bills-with-freeagent/113813)
**Author:** [@ellwould](https://community.asterisk.org/u/ellwould)\ **Replies:** 4\ **Last updated:** [September 24, 2026, 12:12pm UTC](https://community.asterisk.org/t/wrote-an-asterisk-gui-that-automatically-bills-with-freeagent/113813 "2026-09-24T12:12:23Z")
YAP (Yet Another PBX) is now v1.0.0 License AGPLv3 For a while I had wanted to write a telephone system that would also handle billing and interconnect with 3rd party accountancy software. After 10 months working on it… --- ## [YAP (Yet Another PBX) can now bill Asterisk and Carrier CDRs using CDR Importer](https://community.asterisk.org/t/yap-yet-another-pbx-can-now-bill-asterisk-and-carrier-cdrs-using-cdr-importer/114050)
**Author:** [@ellwould](https://community.asterisk.org/u/ellwould)\ **Replies:** 3\ **Last updated:** [September 24, 2026, 12:01pm UTC](https://community.asterisk.org/t/yap-yet-another-pbx-can-now-bill-asterisk-and-carrier-cdrs-using-cdr-importer/114050 "2026-09-24T12:01:07Z")
Wrote a program to import Asterisk and carrier CDRs, re-rate them, insert them into YAP (Yet Another PBX) First stable release of CDR Importer v1.0.0 It is part of the YAP (Yet Another PBX) open source project and like… --- ## [TLS configuration and budget-friendly phones (Denwa/XUE)](https://community.asterisk.org/t/tls-configuration-and-budget-friendly-phones-denwa-xue/114404)
**Author:** [@dti.voip](https://community.asterisk.org/u/dti.voip)\ **Replies:** 1\ **Last updated:** [September 23, 2026, 12:01pm UTC](https://community.asterisk.org/t/tls-configuration-and-budget-friendly-phones-denwa-xue/114404 "2026-09-23T12:01:20Z")
Hi everyone, I am setting up Asterisk v22. I have already configured my trunk and the endpoints which include Grandstream GXP1628 phones, but mostly Denwa DW210P and XUE S3 models. Currently, they are operating via UDP,… --- ## [What happened to asterisksounds.org?](https://community.asterisk.org/t/what-happened-to-asterisksounds-org/114399)
**Author:** [@gsaelzbrot](https://community.asterisk.org/u/gsaelzbrot)\ **Replies:** 6\ **Last updated:** [September 22, 2026, 7:34pm UTC](https://community.asterisk.org/t/what-happened-to-asterisksounds-org/114399 "2026-09-22T19:34:51Z")
Hello all. I’m quite new to Asterisk and I’ve set up a system for home use, which has been working great so far. However, I’d now like to use German sound files. I’ve found some information indicating that you can downlo… [Next page](https://community.asterisk.org/latest.md?page=1)