Asterisk
| Topic | Replies | Views | Activity | |
|---|---|---|---|---|
| Asterisk Internal Call Failures: Reasons and How to Fix Them |
|
0 | 25 | November 25, 2025 |
| No sound over call |
|
2 | 20 | November 25, 2025 |
| Asterisk stops receiving calls after several minutes |
|
8 | 35 | November 24, 2025 |
| In the rtp.conf.sample i saw the Stun url and turn address url, need some knowledge about that |
|
3 | 26 | November 24, 2025 |
| Asterisk dealing with ATA HT813 |
|
6 | 56 | November 24, 2025 |
| Hangup cause as unknown in case of timeout |
|
3 | 37 | November 22, 2025 |
| Combining Dial G() and b() |
|
2 | 21 | November 21, 2025 |
| Explicit Call Transfer (ECT) or Facility Messages for call transfers over EuroISDN |
|
4 | 31 | November 21, 2025 |
| Asterisk SIP Sample Messages/ Unit Tests |
|
3 | 32 | November 21, 2025 |
|
Asterisk Release 22.7.0
The Asterisk Development Team would like to announce the release of asterisk-22.7.0. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The official Asterisk P… |
|
2 | 41 | November 20, 2025 |
|
Asterisk Release 21.12.0
The Asterisk Development Team would like to announce the release of asterisk-21.12.0. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The official Asterisk … |
|
2 | 23 | November 20, 2025 |
|
Asterisk Release 23.1.0
The Asterisk Development Team would like to announce the release of asterisk-23.1.0. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The official Asterisk P… |
|
2 | 91 | November 20, 2025 |
|
Asterisk Release 20.17.0
The Asterisk Development Team would like to announce the release of asterisk-20.17.0. The release artifacts are available for immediate download at and Repository: GitHub - asterisk/asterisk: The official Asterisk … |
|
2 | 27 | November 20, 2025 |
| Queue Ordered Ring Strategy |
|
2 | 28 | November 20, 2025 |
| Voice delay, wrong buffering |
|
4 | 47 | November 20, 2025 |
| I had create an Android app for Voip SIP calls with Asterisk Server |
|
3 | 86 | November 20, 2025 |
| How to get values of queue_log variables |
|
0 | 11 | November 19, 2025 |
| Does Asterisk Support the External Media over a webRTC |
|
1 | 25 | November 19, 2025 |
| Asterisk side of Hotdesking |
|
3 | 56 | November 19, 2025 |
| outgoing calls not working with sip trunk with tls PJSIP |
|
8 | 41 | November 18, 2025 |
| Manager.conf eventfilter syntax |
|
4 | 29 | November 18, 2025 |
| Extension not register |
|
2 | 18 | November 18, 2025 |
| Pjsip callerid= specification -- syntax? |
|
3 | 39 | November 18, 2025 |
| Direct media setup didn't sending the Re-INVITE |
|
7 | 27 | November 17, 2025 |
| Asterisk sends two SDPs after negotiation is done |
|
4 | 43 | November 15, 2025 |
| Call Hangup Issue |
|
1 | 25 | November 14, 2025 |
| Is it possible to track active Playback so I can interrupt it? |
|
14 | 97 | November 13, 2025 |
| Call Transfer Issue — Asterisk Not Reaching SIP User on call center |
|
6 | 50 | November 13, 2025 |
| Migrating from Asterisk 16 → Asterisk 21 – challenges, changes, and precautions |
|
1 | 53 | November 13, 2025 |
| Task processor stasis/p:manager:core orphaned after console reload in Asterisk 22.5 |
|
4 | 39 | November 13, 2025 |