# Asterisk Support **URL:** https://community.asterisk.org/c/asterisk/asterisk-support/13.md [Latest](https://community.asterisk.org/latest.md) · [Categories](https://community.asterisk.org/categories.md) --- ## [About the Asterisk Support category](https://community.asterisk.org/t/about-the-asterisk-support-category/19)
**Author:** [@system](https://community.asterisk.org/u/system)\ **Replies:** 1\ **Last updated:** [February 5, 2025, 9:21am UTC](https://community.asterisk.org/t/about-the-asterisk-support-category/19 "2025-02-05T09:21:56Z")
If you are having difficulties installing, operating, upgrading or configuring Asterisk, post your issues in Support. Please make sure to check resources and to search the forum before posting. Most standard and basic qu… --- ## [Asterisk-GUI](https://community.asterisk.org/t/asterisk-gui/48774)
**Author:** [@malcolmd](https://community.asterisk.org/u/malcolmd)\ **Replies:** 0\ **Last updated:** [April 29, 2015, 2:09pm UTC](https://community.asterisk.org/t/asterisk-gui/48774 "2015-04-29T14:09:51Z")
Howdy, Asterisk-GUI is being pulled from Digium and Asterisk download servers. It is a dead GUI project that has not been updated in more than 4 years. It has received no bug fixes or enhancements in this time, and i… --- ## [Troubleshooting WebRTC Issues](https://community.asterisk.org/t/troubleshooting-webrtc-issues/46011)
**Author:** [@navaismo](https://community.asterisk.org/u/navaismo)\ **Replies:** 8\ **Last updated:** [August 30, 2018, 9:16am UTC](https://community.asterisk.org/t/troubleshooting-webrtc-issues/46011 "2018-08-30T09:16:46Z")
Here is a little guide to troubleshoot webrtc issues with Asterisk. Always try to use the latest WebRTC API with the latest Asterisk branch(11 or 12). Be sure you have the icessuport enabled in the rtp.conf ISSUE: I … --- ## [Telephony Card Support](https://community.asterisk.org/t/telephony-card-support/38386)
**Author:** [@malcolmd](https://community.asterisk.org/u/malcolmd)\ **Replies:** 0\ **Last updated:** [February 27, 2012, 2:36pm UTC](https://community.asterisk.org/t/telephony-card-support/38386 "2012-02-27T14:36:32Z")
Hey!! You've got a Digium telephony card and you need Support? Digium doesn't provide support via these forums, you have to contact the Support department directly. Contact Digium's Support department - digium.com/s… --- ## [Recurring Asterisk segfaults in res\_http\_websocket.so (error 4)](https://community.asterisk.org/t/recurring-asterisk-segfaults-in-res-http-websocket-so-error-4/114859)
**Author:** [@Diego1990](https://community.asterisk.org/u/Diego1990)\ **Replies:** 3\ **Last updated:** [October 28, 2026, 1:44pm UTC](https://community.asterisk.org/t/recurring-asterisk-segfaults-in-res-http-websocket-so-error-4/114859 "2026-10-28T13:44:15Z")
Hi everyone, (Note: English is not my native language; this post was translated from Spanish using Gemini). We are experiencing repeated crashes/segfaults in Asterisk in our production environment. Looking into /var/lo… --- ## [Asterisk WebSocket](https://community.asterisk.org/t/asterisk-websocket/114853)
**Author:** [@leandroarbia](https://community.asterisk.org/u/leandroarbia)\ **Replies:** 2\ **Last updated:** [October 22, 2026, 1:33pm UTC](https://community.asterisk.org/t/asterisk-websocket/114853 "2026-10-22T13:33:41Z")
We investigated an issue in Asterisk 22.9.0 where WebSocket connectivity stopped working and the HTTP server reported: HTTP session count exceeded 1000 sessions. Restarting Asterisk temporarily restored service. The in… --- ## [app\_voicemail\_imap failure, Asterisk 22.7.0 local build, stopped working after OS upgrade and Asterisk rebuild](https://community.asterisk.org/t/app-voicemail-imap-failure-asterisk-22-7-0-local-build-stopped-working-after-os-upgrade-and-asterisk-rebuild/114823)
**Author:** [@trounds](https://community.asterisk.org/u/trounds)\ **Replies:** 2\ **Last updated:** [October 20, 2026, 3:29pm UTC](https://community.asterisk.org/t/app-voicemail-imap-failure-asterisk-22-7-0-local-build-stopped-working-after-os-upgrade-and-asterisk-rebuild/114823 "2026-10-20T15:29:19Z")
I recently upgraded my home server from Ubuntu 24.04 LTS to Ubuntu 26.04 LTS. Under 24.04, I ran Asterisk 22.7.0 (which I built locally) without issue. The relevant functionality was to pick up SIP calls (from an Obihai … --- ## [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:** 2\ **Last updated:** [October 16, 2026, 12:56pm 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 "2026-10-16T12:56:05Z")
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… --- ## [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… --- ## [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… --- ## [Downloads.digium.com is missing a directory](https://community.asterisk.org/t/downloads-digium-com-is-missing-a-directory/114376)
**Author:** [@miken32](https://community.asterisk.org/u/miken32)\ **Replies:** 2\ **Last updated:** [September 18, 2026, 4:58pm UTC](https://community.asterisk.org/t/downloads-digium-com-is-missing-a-directory/114376 "2026-09-18T16:58:52Z")
https://downloads.digium.com/pub/telephony/codec\_siren14/ seems to be missing the asterisk-24.0 directory. It’s present in the other third party codec directories. --- ## [Queue ringall add additional members](https://community.asterisk.org/t/queue-ringall-add-additional-members/114356)
**Author:** [@PitzKey](https://community.asterisk.org/u/PitzKey)\ **Replies:** 3\ **Last updated:** [September 17, 2026, 7:17am UTC](https://community.asterisk.org/t/queue-ringall-add-additional-members/114356 "2026-09-17T07:17:19Z")
Hello, I’m running Asterisk 20.20.1 with a ringall queue where members have different penalties. The goal is to widen the ring as the caller waits: penalty 0 rings immediately, penalty 1 joins after 10 seconds, penalty … --- ## [How do you normally handle empty files created by MixMonitor(b)?](https://community.asterisk.org/t/how-do-you-normally-handle-empty-files-created-by-mixmonitor-b/114345)
**Author:** [@benphone](https://community.asterisk.org/u/benphone)\ **Replies:** 5\ **Last updated:** [September 16, 2026, 9:04pm UTC](https://community.asterisk.org/t/how-do-you-normally-handle-empty-files-created-by-mixmonitor-b/114345 "2026-09-16T21:04:14Z")
How do you normally handle empty files created by MixMonitor(b)? When using: MixMonitor(filename.wav,b) Asterisk creates the recording file immediately, even if the channel is never bridged. The b option prevents aud… --- ## [High RSS / OOM after heavy AMI Originate campaigns — looks like glibc arena fragmentation, not a leak](https://community.asterisk.org/t/high-rss-oom-after-heavy-ami-originate-campaigns-looks-like-glibc-arena-fragmentation-not-a-leak/114243)
**Author:** [@benphone](https://community.asterisk.org/u/benphone)\ **Replies:** 7\ **Last updated:** [September 15, 2026, 3:05pm UTC](https://community.asterisk.org/t/high-rss-oom-after-heavy-ami-originate-campaigns-looks-like-glibc-arena-fragmentation-not-a-leak/114243 "2026-09-15T15:05:06Z")
We run big outbound campaigns via concurrent AMI Originate (peaked around 763 channels / 1200 threads, RSS ~41 GiB). After the campaign drains and threads exit, RSS stays around ~27 GiB with only 68 threads left. Grabbe… --- ## [Asterisk Essentials Course](https://community.asterisk.org/t/asterisk-essentials-course/114198)
**Author:** [@PitzKey](https://community.asterisk.org/u/PitzKey)\ **Replies:** 2\ **Last updated:** [September 8, 2026, 9:26pm UTC](https://community.asterisk.org/t/asterisk-essentials-course/114198 "2026-09-08T21:26:46Z")
Hello, Is there an updated one that covers PJSIP? Thank you --- ## [Understanding what's causing load/CPU spike](https://community.asterisk.org/t/understanding-whats-causing-load-cpu-spike/114144)
**Author:** [@Dovid](https://community.asterisk.org/u/Dovid)\ **Replies:** 6\ **Last updated:** [September 7, 2026, 6:47am UTC](https://community.asterisk.org/t/understanding-whats-causing-load-cpu-spike/114144 "2026-09-07T06:47:47Z")
Hi, I use MusicOnHold + Realtime extensively with Asterisk. Since I am using realtime once a class is added it stays in memory until I un-regiser the MOH class. To do this I added to the dialplan a script that checks… --- ## [Build errors on 22.11](https://community.asterisk.org/t/build-errors-on-22-11/114122)
**Author:** [@miken32](https://community.asterisk.org/u/miken32)\ **Replies:** 2\ **Last updated:** [September 1, 2026, 10:44am UTC](https://community.asterisk.org/t/build-errors-on-22-11/114122 "2026-09-01T10:44:49Z")
My builds which were successful in 22.10.1 are now failing during the pjproject configure script. I assume this is related to commit 9778364 but I’m not knowledgeable enough to know if it’s something I can/should resolve… --- ## [No early media on multiple device call](https://community.asterisk.org/t/no-early-media-on-multiple-device-call/113993)
**Author:** [@rashadmlk](https://community.asterisk.org/u/rashadmlk)\ **Replies:** 2\ **Last updated:** [August 24, 2026, 11:12am UTC](https://community.asterisk.org/t/no-early-media-on-multiple-device-call/113993 "2026-08-24T11:12:48Z")
I have hikvision indoor monitor, linphone, and webrtc device (mobile app) that sits on same endpoint. That has to get early media simultaneously. The problem is after registering hikvision to server. I don’t see any earl… --- ## [Menuselect enabled option disappears](https://community.asterisk.org/t/menuselect-enabled-option-disappears/113955)
**Author:** [@miken32](https://community.asterisk.org/u/miken32)\ **Replies:** 2\ **Last updated:** [August 19, 2026, 5:42pm UTC](https://community.asterisk.org/t/menuselect-enabled-option-disappears/113955 "2026-08-19T17:42:25Z")
I’m having some trouble building Asterisk with ODBC voicemail module. This is a non-interactive install, all scripted. The module seems to be enabled when I ask for it but then is again disabled with the next run of menu… --- ## [PJSIP bundled doesn't fallback to A record after NAPTR/SRV lookup failure](https://community.asterisk.org/t/pjsip-bundled-doesnt-fallback-to-a-record-after-naptr-srv-lookup-failure/113763)
**Author:** [@vgfaria](https://community.asterisk.org/u/vgfaria)\ **Replies:** 3\ **Last updated:** [August 3, 2026, 5:36pm UTC](https://community.asterisk.org/t/pjsip-bundled-doesnt-fallback-to-a-record-after-naptr-srv-lookup-failure/113763 "2026-08-03T17:36:08Z")
Hi everyone, I’m facing an issue with the bundled PJPROJECT in Asterisk and I’d like to know if anyone has seen this behavior before. Environment Asterisk 18.24.3 Bundled PJPROJECT (2.14.1) Alpine Linux OpenSSL 3 Tran… --- ## [Bug Report: pjsip/outsess Taskprocessors Persist After Call Completion](https://community.asterisk.org/t/bug-report-pjsip-outsess-taskprocessors-persist-after-call-completion/113754)
**Author:** [@vatss90](https://community.asterisk.org/u/vatss90)\ **Replies:** 9\ **Last updated:** [July 30, 2026, 4:08pm UTC](https://community.asterisk.org/t/bug-report-pjsip-outsess-taskprocessors-persist-after-call-completion/113754 "2026-07-30T16:08:58Z")
Summary On Asterisk 22.10.1, pjsip/outsess taskprocessors continue to accumulate after outbound PJSIP calls have completed. The taskprocessors remain registered even when there are no active outbound calls. While invest… --- ## [Asterisk getting locked randomly](https://community.asterisk.org/t/asterisk-getting-locked-randomly/113752)
**Author:** [@PitzKey](https://community.asterisk.org/u/PitzKey)\ **Replies:** 3\ **Last updated:** [July 29, 2026, 3:51pm UTC](https://community.asterisk.org/t/asterisk-getting-locked-randomly/113752 "2026-07-29T15:51:26Z")
Hello, Please see attached backtrace asterisk-wedge-bt-1041.txt (204.1 KB) From what I can tell: A thread completing a call transfer and a thread doing a routine RTP maintenance check each grabbed one lock and are wai… --- ## [Error pjsip taskprocessors got stuck why](https://community.asterisk.org/t/error-pjsip-taskprocessors-got-stuck-why/113696)
**Author:** [@benphone](https://community.asterisk.org/u/benphone)\ **Replies:** 5\ **Last updated:** [July 22, 2026, 2:43pm UTC](https://community.asterisk.org/t/error-pjsip-taskprocessors-got-stuck-why/113696 "2026-07-22T14:43:26Z")
\[root@pbx5IL asterisk\]# grep pjproject full.1 \[2026-07-22 04:49:54\] WARNING\[50450\] pjproject: sip\_transport.c Dropping 154 bytes packet from TCP 81.199.227.177:25247 : PJSIP syntax error exception when par… --- ## [PickupChan not working](https://community.asterisk.org/t/pickupchan-not-working/113477)
**Author:** [@benphone](https://community.asterisk.org/u/benphone)\ **Replies:** 13\ **Last updated:** [July 20, 2026, 9:07am UTC](https://community.asterisk.org/t/pickupchan-not-working/113477 "2026-07-20T09:07:51Z")
looks like something change with pickupchan PickupChan("PJSIP/200-GIS-000dc8f8", "PJSIP/203-GIS,p") but for some reason it does not get bridged do not see a obvious error i think it has to do with the last asterisk 2… --- ## [Asterisk 22.8.2 hangs during ast\_hangup() - AO2 channel container lock contention causes new calls to stop](https://community.asterisk.org/t/asterisk-22-8-2-hangs-during-ast-hangup-ao2-channel-container-lock-contention-causes-new-calls-to-stop/113416)
**Author:** [@vatss90](https://community.asterisk.org/u/vatss90)\ **Replies:** 9\ **Last updated:** [July 3, 2026, 12:08pm UTC](https://community.asterisk.org/t/asterisk-22-8-2-hangs-during-ast-hangup-ao2-channel-container-lock-contention-causes-new-calls-to-stop/113416 "2026-07-03T12:08:37Z")
Asterisk Version : 22.8.2 OS : RHEL 9.7 Kernel : 5.14.0-611.54.1.el9\_7.x86\_64 Deployment : Podman Container CPU : 256 cores Workload : Incoming call platform ARI … --- ## [Delay in asterisk sending outbound invite](https://community.asterisk.org/t/delay-in-asterisk-sending-outbound-invite/113290)
**Author:** [@alok](https://community.asterisk.org/u/alok)\ **Replies:** 7\ **Last updated:** [June 22, 2026, 9:42am UTC](https://community.asterisk.org/t/delay-in-asterisk-sending-outbound-invite/113290 "2026-06-22T09:42:15Z")
My local agents while sending an invite to asterisk get an immediate response of trying but asterisk delays the invite to outgoing sip trunk by more than 10 seconds. Asterisk version is 20.19 --- ## [Question regarding wrapuptime behavior after call disposition in Queue](https://community.asterisk.org/t/question-regarding-wrapuptime-behavior-after-call-disposition-in-queue/113322)
**Author:** [@Mehwish](https://community.asterisk.org/u/Mehwish)\ **Replies:** 2\ **Last updated:** [June 20, 2026, 6:29am UTC](https://community.asterisk.org/t/question-regarding-wrapuptime-behavior-after-call-disposition-in-queue/113322 "2026-06-20T06:29:18Z")
Hello, I am currently working on a call center system using Asterisk queues and need clarification regarding the wrapuptime behavior. Current System Behavior In my current setup: A campaign is running and calls are b… --- ## [Local channel having no idea about call id and for that matter any such variables](https://community.asterisk.org/t/local-channel-having-no-idea-about-call-id-and-for-that-matter-any-such-variables/113312)
**Author:** [@devintern123](https://community.asterisk.org/u/devintern123)\ **Replies:** 2\ **Last updated:** [June 18, 2026, 10:23am UTC](https://community.asterisk.org/t/local-channel-having-no-idea-about-call-id-and-for-that-matter-any-such-variables/113312 "2026-06-18T10:23:39Z")
ISSUE IN DETAIL specific case : call transfer (NB: i cant change existing architecture) When Agent 1 transfers a call to Agent 2, the SIP protocol handles this via a REFER message. Asterisk receives this and must tear … --- ## [FIFO for multiple Queues](https://community.asterisk.org/t/fifo-for-multiple-queues/113288)
**Author:** [@w1ntermute](https://community.asterisk.org/u/w1ntermute)\ **Replies:** 2\ **Last updated:** [June 18, 2026, 8:41am UTC](https://community.asterisk.org/t/fifo-for-multiple-queues/113288 "2026-06-18T08:41:51Z")
Cheers everyone, ive been testing and searching for a way to achive FIFO within multiple Queues that an agent is part of? So Agent A ist Part of Queue 1 and 2 all agents of these Queues are in Calls Caller X is waiti… --- ## [Activation/Enhancement of the Jitter Buffer in PJSIP](https://community.asterisk.org/t/activation-enhancement-of-the-jitter-buffer-in-pjsip/113297)
**Author:** [@Ahoumenou](https://community.asterisk.org/u/Ahoumenou)\ **Replies:** 3\ **Last updated:** [June 17, 2026, 6:31pm UTC](https://community.asterisk.org/t/activation-enhancement-of-the-jitter-buffer-in-pjsip/113297 "2026-06-17T18:31:39Z")
Hello everyone, By default, PJSIP appears to enable a jitter buffer with relatively low values, which may not be sufficient in environments where network quality is unstable. In this context, we are considering strengt… [Next page](https://community.asterisk.org/c/asterisk/asterisk-support/13.md?page=1)