Changelog

New Features & Improvements

1. Add ability to override certain TTS streaming options via the config verb, allowing runtime control of streaming behavior.
2. Add ability to enable/disable Azure audio logging via azureOptions in speech credentials.
3. Compare SDP to determine if transcoding is being used, with refactored codec checking.
4. SoundHound now supports audio endpoint configuration from speech credentials with requestInfo and sampleRate options.
5. Add configurable say chunk size for TTS streaming.
6. Enhanced TTS sentence boundary detection for Arabic and Japanese with improved regex handling.
7. Add support for sending DTMF to Ultravox.
8. Add label to STT/TTS alerts with time-series updates.
9. Use timeout on HTTP requests to prevent hanging connections.
10. Allow UAS leg to send re-invite with outbound gateway credentials.
11. Add hasRecording flag for setting recording URLs in call detail records.
12. Add configurable backup for outbound registration failure.
13. Use persistent call-id for regbot using gateway SID.
14. Only disable registration after multiple consecutive failures rather than a single failure.
15. Add support for Google Gemini TTS.
16. Add support for OpenAI transcribe auto language detection.
17. Allow boostAudioSignal from updateCall API.
18. Allow media_path updates from REST API with validation for media_path values.
19. Allow startRecording without SIPREC URL for cloud deployments.
20. Add new fields for ICE and DTLS configuration.
21. Add admin carrier and number management control via JAMBONES_ADMIN_CARRIER.
22. Add database migrations for predefined carriers tables.
23. Add alert notification badge to the webapp with configurable polling.
24. Add SoundHound audio endpoint configuration in the webapp.

Bug fixes

• Fixed say verb does not close streaming when finishing say.
• Fixed playbackIds not in correct order compared with say.text array.
• Allow say verb to fail as NonFatalTaskError for File Not Found instead of crashing the call.
• Fixed race condition in gather where timeout timer gets set after resolve when speech transcript arrives.
• Fixed transcribe 2 legs cannot fallback.
• Fixed SDP checking for Opus on A leg during B leg dialing.
• Fixed undefined issue when setting TTS streaming channel vars.
• Fixed dial verb cannot bridge 2 leg endpoints due to transcoding.
• Do not send TTS streaming events when not doing TTS streaming.
• Fixed gather should ignore transcription if task is killed or resolved.
• Fixed callsession cannot close TTS streaming.
• Optimized slow SQL queries and added support for readonly database endpoints.
• Optimized CIDR gateway queries using STRAIGHT_JOIN to prevent inefficient table scans.
• Parallelized independent gateway queries for exact IP and CIDR range lookups to reduce inbound call routing latency.
• When far end answers with only PCMA, passthrough instead of transcoding to PCMU.
• Include codec-accept on answer to rtpengine during reinvites.
• Remove ICE and DTLS settings if set in database configuration.
• Fixed reg trunk proxy port handling to only use port if proxy is IPv4 address; use SIP gateway host for DNS lookup of ephemeral gateways.
• Fixed Array.prototype.push.apply stack overflow causing maximum call stack size exceeded errors.
• If no SRV record found when no port specified, fall back to A record lookup.
• Fixed support for writer/reader database nodes.
• Removed send_options_bots array to fix ticket #1983.
• Fixed timing issue with ephemeral gateway update/deletion and potential regbot zombie processes.
• Set JAMBONES_NETWORK_CIDR as private IP address space when running under K8S.
• Removed activation code from API response.
• Force account sip_realm to lowercase.
• Fixed obscured key detection.
• Fixed unable to fetch voice_call_session.
• Fixed webhook URL validation on update to not remove URLs if unchanged.
• Fixed Soniox STT speech credential validation.
• Subscription update-quantities validation for minimum voice call sessions.
• Added database indexes and SQL optimizations including index on sip_gateways and trunk_type for predefined carriers.
• Fixed duplicate inbound and outbound SIP gateways could be created.
• Require IP auth trunk to have either inbound or outbound carrier; consistent wording to “IP Trunk”.
• Fixed outbound call routing race condition showing default LCR route set that user cannot delete.
• Removed filter for active carriers from phone number configuration.

SQL changes

ALTER TABLE sip_gateways ADD COLUMN remove_ice BOOLEAN NOT NULL DEFAULT 0
ALTER TABLE sip_gateways ADD COLUMN dtls_off BOOLEAN NOT NULL DEFAULT 0
CREATE INDEX idx_sip_gateways_inbound_lookup ON sip_gateways (inbound,netmask,ipv4)
CREATE INDEX idx_sip_gateways_inbound_netmask ON sip_gateways (inbound,netmask)
ALTER TABLE predefined_carriers ADD COLUMN trunk_type ENUM('static_ip','auth','reg') NOT NULL DEFAULT 'static_ip'
ALTER TABLE predefined_sip_gateways ADD COLUMN send_options_ping BOOLEAN NOT NULL DEFAULT 0
ALTER TABLE predefined_sip_gateways ADD COLUMN use_sips_scheme BOOLEAN NOT NULL DEFAULT 0
ALTER TABLE predefined_sip_gateways ADD COLUMN pad_crypto BOOLEAN NOT NULL DEFAULT 0
ALTER TABLE predefined_sip_gateways ADD COLUMN remove_ice BOOLEAN NOT NULL DEFAULT 0
ALTER TABLE predefined_sip_gateways ADD COLUMN dtls_off BOOLEAN NOT NULL DEFAULT 0
ALTER TABLE predefined_sip_gateways ADD COLUMN protocol ENUM('udp','tcp','tls', 'tls/srtp') NOT NULL DEFAULT 'udp'

Availability

• Available now with devops scripts for subscription customers

Questions? Contact us at support@jambonz.org