One-way VoIP audio
Trace the missing direction from advertised media address to receiving endpoint.
Asterisk 22 LTS · FreePBX 17 / Debian 12 checksFind your next answer
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24 results for “one-way audio”
Intermittent one-way audio after hold, transfer or idle time
Compare the exact working and failing call phase, SDP direction and RTP path across hold, resume, transfer or idle time before changing NAT, direct-media or timeout controls.
RFC 3264 offer/answer behavior, current Asterisk res_pjsip media controls and Wireshark RTP analysis documentation reviewed October 5, 2026; no PBX, endpoint, trunk, firewall, packet stream, media transition or call testedFreePBX call connects but has one-way or no audio
Trace the final SDP and observed RTP through both FreePBX call legs before changing NAT, direct media, symmetric RTP, Strict RTP, codecs or security.
Sangoma PBX GUI Asterisk SIP Settings and legacy FreePBX 12 SIP-audio guidance plus current Asterisk res_pjsip NAT, configuration, troubleshooting, CHANNEL and bounded-debug references reviewed October 6, 2026; the legacy Sangoma page is chan_sip-era symptom and settings context, not a FreePBX 17 PJSIP configuration recipe, and no PBX, firewall, endpoint, trunk, packet flow or live call was inspected or testedVoIPstudio Desktop is online but the call or audio fails
Separate application login, endpoint registration, call signaling, selected devices, WebRTC reachability and two-way media before changing credentials or network policy.
Current VoIPstudio download, Desktop Softphone, user-introduction, endpoint-administration and network-troubleshooting documentation reviewed October 16, 2026; the public download surface lists Windows, Windows MSI, macOS Intel, macOS Silicon and Linux paths but does not publish one common desktop build number, and no package, workstation, operating-system permission, user account, endpoint record, VPN, firewall, WebRTC test, SIP Test, MTR result, packet, protected session, media stream or live call was inspected or testedVoIPstudio Desktop Softphone: account, audio and network evidence
Evaluate the service-specific Windows, macOS and Linux desktop client without treating an online icon, selected device or WebRTC test as proof of a completed two-way call.
Current VoIPstudio download, Desktop Softphone, user-introduction, endpoint-administration and network-troubleshooting documentation reviewed October 16, 2026; the public download surface lists Windows, Windows MSI, macOS Intel, macOS Silicon and Linux paths but does not publish one common desktop build number, and no package, workstation, operating-system permission, user account, endpoint record, VPN, firewall, WebRTC test, SIP Test, MTR result, packet, protected session, media stream or live call was inspected or testedAlgo 8201 IP PoE door intercom
Evaluate the 8201 as a one-extension SIP door intercom whose call button, relay, sensor, multicast, outdoor installation and management paths require separate acceptance.
Algo · Outdoor SIP door and room intercomAlgo 8211 IP Single Button
Evaluate the 8211 as a standalone PoE button with separate press and long-press actions, outbound prerecorded SIP audio, multicast, API and supervision boundaries.
Algo · Standalone PoE single call and alert buttonFanvil V62 Pro: Bluetooth handset and business-phone boundaries
Evaluate the V62 Pro while keeping its BTH60 Bluetooth handset, SIP identities, DSS controls, power, headset paths, firmware and service acceptance independently verifiable.
Fanvil · SIP desk phone with Bluetooth handsetFanvil V64: Wi-Fi, Bluetooth and DSS boundaries
Evaluate the V64 as a 12-line desk phone while keeping SIP identities, DSS assignments, active calls, radios, ports, power, firmware and PBX-dependent features separate.
Fanvil · Business SIP desk phone with built-in Wi-Fi and BluetoothFanvil V66 Pro: touch, video preview and platform-document boundaries
Evaluate the V66 Pro while keeping its touch interface, BTH60 handset, receive-side video, Wi-Fi, Bluetooth, platform-document conflict, firmware and call service exact.
Fanvil · Touch-screen SIP desk phone with Bluetooth handset and receive-side videoFanvil V67: Android video-phone, camera and application boundaries
Evaluate the V67 as an Android SIP video desk phone while keeping its 20 identities, touch interface, camera, applications, DSS assignments, conferencing, radios, security controls and firmware separate.
Fanvil · Android SIP video desk phone with adjustable touch display, built-in Wi-Fi and BluetoothFreePBX paging and intercom deployment
Choose an explicit audience, verify endpoint auto-answer and distinguish broadcast delivery from two-way intercom.
FreePBX 17 paging-module source plus Sangoma's version-13-labelled Paging and Intercom guide and current Sangoma phone header documentation reviewed September 29, 2026; no PBX, page group, endpoint, header, firmware, audio path or live page was testedGrandstream GAC2570 conference phone
Separate the room's SIP, meeting-control and BYOD audio paths before accepting microphone reach, mobility or conferencing behavior.
Grandstream · Conference phone and room-audio endpointGrandstream GHP630W white Wi-Fi 6 hotel phone
Assess the white GHP630W as a Wi-Fi 6 or PoE guest-room SIP phone without transferring black-model, wired-only or package details from its siblings.
Grandstream · Wi-Fi 6 and PoE guest-room SIP phoneGrandstream GHP631 black wired hotel IP phone
Assess the black, wired-only GHP631 by its PoE path, color display and room-service workflow without importing Wi-Fi or local-adapter evidence from W siblings.
Grandstream · Wired PoE guest-room SIP phoneGrandstream GHP631W black Wi-Fi 6 hotel phone
Assess the black GHP631W as a Wi-Fi 6 or PoE guest-room SIP phone while preserving its exact color, radio, power, conference and package boundaries.
Grandstream · Wi-Fi 6 and PoE guest-room SIP phoneGrandstream GSC3506 v2 multicast paging speaker
Evaluate the GSC3506 v2 as a microphone-free 30 W SIP and multicast paging speaker with PoE++ input, analog output and sensor-trigger boundaries.
Grandstream · SIP and multicast paging speakerGrandstream GSC3510 SIP intercom speaker
Plan a two-way indoor paging endpoint with SIP, multicast, PoE and microphone privacy boundaries.
Grandstream · SIP intercom and paging speakerGrandstream GSC3570 facility control station
Evaluate the GSC3570 as a four-account SIP intercom and facility-control panel while keeping video, alarm I/O, relay ownership, storage, firmware and physical safety independently observable.
Grandstream · HD intercom and facility control stationVoIP troubleshooting
Use symptoms and evidence to isolate registration, routing, media and device faults.
FoundationalYealink CP925 conference phone
Evaluate the CP925 as a SIP conference phone with local conferencing, separate USB or Bluetooth BYOD paths and an optional analog PSTN accessory; treat published pickup and security features as capabilities to verify in the intended room and configuration.
Yealink · Conference phoneYealink CP935W Wi-Fi conference phone
Evaluate the CP935W as a battery-powered Wi-Fi SIP conference phone while keeping its registered SIP, Bluetooth, USB host, battery, room-audio and firmware paths separate.
Yealink · Battery-powered Wi-Fi SIP conference phoneYealink CP935W-Base DECT conference system
Evaluate the CP935W-Base as a CP935W conference endpoint attached to a W70B single-cell DECT base, keeping endpoint, base, account, radio, host-audio and battery limits distinct.
Yealink · Battery-powered conference phone with W70B single-cell DECT baseYealink CP965
Evaluate the current CP965 as an exact SIP conference-phone model, keeping its room pickup, one SIP account, local conference, hybrid-host paths, expansion microphones, PoE and firmware lane separate.
Yealink · Conference phone