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fix/missing-app-sid-and-trace
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Author SHA1 Message Date
rhonda hollis f567049bfb refactor to remove Redis 
- prevent drachtio from auto passing the non-success response
 2026-05-18 09:06:24 -07:00
rhonda hollis ac7f8c4424 ensure traceId is set on CallRecord 2026-05-14 11:43:47 -07:00
Hoan Luu Huu cb581f9be7 support recognizer.googleOptions.parentPath (#1546) 
* support recognizer.googleOptions.parentPath

* update vẻb specification

* wip
 2026-05-13 08:33:51 -04:00
Hoan Luu Huu 423c070900 fixed HTTP llm.toolHook does not support openAI (#1547) 2026-05-06 08:47:04 -07:00
Hoan Luu Huu 4b8fc65cdb support houndify ws (#1540) 0.9.7-rc1 v0.9.7-rc1 2026-04-23 07:19:39 -04:00
Hoan Luu Huu 570641fe07 update drachtio srf 5.0.21 (#1537) 2026-04-13 21:52:14 -04:00
Dave Horton 4f373d2fbc bump version v0.9.6 2026-03-31 07:43:27 -04:00
Sam Machin 24d9740618 add x-reason to sip-decline (#1518) 
* add x-ver to sip-decline

* lint

* Update sip_decline.js

---------

Co-authored-by: Dave Horton <daveh@beachdognet.com>
0.9.6 		2026-03-29 16:19:15 -04:00
rhondahollis 39746598b5 add null check for eventHook (#1534) 
* add null check for eventHook

* move guard to superclass, remove logging that adds no value

---------

Co-authored-by: rhonda hollis <rhonda@jambonz.org>
Co-authored-by: Dave Horton <daveh@beachdognet.com>
 2026-03-29 16:13:35 -04:00
Sam Machin 315eb98d86 add sp_sid to alerts (#1533) 
* add sp_sid to alerts

* bump time-series

---------

Co-authored-by: Dave Horton <daveh@beachdognet.com>
 2026-03-29 16:07:08 -04:00
Dave Horton df30496dac fix uncaught exception referencing this.ep in freeswitch hangup scenario (#1532) 2026-03-27 08:31:32 -04:00
Sam Machin 5d6751782a Fix/hangup call (#1530) 
* Update error.js

* Update error.js
 2026-03-26 08:20:28 -04:00
rhondahollis 6147ec3f6a ensure sbcCallId is added to callInfo (#1529) 
Co-authored-by: rhonda hollis <rhonda@jambonz.org>
 2026-03-25 17:00:05 -04:00
Ed Robbins 18a13971ca respond to re-INVITE during race condition. (#1527) v0.9.6-rc6 2026-03-20 10:41:07 -04:00
Dave Horton 5bd1c53f7d fixed faulty commit https://github.com/jambonz/jambonz-feature-server/pull/1528 2026-03-20 09:17:00 -04:00
Anton Voylenko 5a759791f9 chore: bump node (#1528) 2026-03-20 08:57:16 -04:00
Sam Machin 1f5fa8d49e Update gather.js (#1526) 2026-03-19 15:29:03 -04:00
Anton Voylenko cf0b392c99 Append sip realm for rest dial (#1525) 
* feat: append sip realm for rest dial

* fix: check account sip realm
 2026-03-17 07:30:21 -04:00
Anton Voylenko 68339ced0b fix: conference mute and mute status (#1218) v0.9.6-rc5 2026-03-12 07:55:49 -04:00
Ed Robbins 1560efaf03 remove sensitive data from log statement (#1522) v0.9.6-rc4 2026-03-05 10:04:20 -05:00
Dave Horton 782ce8154e update drachtio-srf (#1521) 2026-03-05 09:33:59 -05:00
Dave Horton 0267acf9e1 anchor media on dial if we are recording (#1520) 2026-02-23 18:13:25 -05:00
rhondahollis 9090006703 update drachtio-srf to v5.0.19 (#1519) 
Co-authored-by: rhonda hollis <rhonda@jambonz.org>
 2026-02-20 16:19:28 -05:00
Hoan Luu Huu d7beaa1b7b Feat/dialogflow cx (#7) (#1516) 
* wip

* wip

* wip

* wip

* logging

* wip

* wip

* wip

* update docker env to latest freeswitch

* wip

* lint

* wip

* support dialogflowcx

* wip

* wip

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---------

Co-authored-by: Dave Horton <daveh@beachdognet.com>
v0.9.6-rc3 		2026-02-12 07:46:36 -05:00
Dave Horton 45d0ca87af update drachtio-srf (#1515) v0.9.6-rc2 2026-02-09 10:31:43 -05:00
Sam Machin bd435dfff9 add deep copy (#1511) 
* escape tag data in listen

* deep copy call data for listen
 2026-02-03 10:44:26 -05:00
Sam Machin b598cd94ae escape tag data in listen (#1510) 2026-02-03 07:35:15 -05:00
Matt Hertogs ceb9a7a3bd Fix boostAudioSignal parameter in Update Call REST API (#1490) 
Corrects the parameter passed to _lccBoostAudioSignal to use
opts.boostAudioSignal instead of the entire opts object, ensuring
the boostAudioSignal option works correctly.

🤖 Generated with [Claude Code](https://claude.com/claude-code)

Co-authored-by: Claude Sonnet 4.5 <noreply@anthropic.com>
v0.9.6-rc1 		2026-01-29 13:58:14 -05:00
Dave Horton ff5f9acaf8 on dial do not reinvite A leg on answer if already answered and we are anchoring media (#1508) 2026-01-29 13:47:53 -05:00
Sam Machin 96cdc2936b invert default (#1507) 2026-01-29 09:22:00 -05:00
Hoan Luu Huu 6120dcbe96 support openai transcribe turn_detection.eagerness (#1496) v0.9.5-11 2026-01-28 08:09:01 -05:00
Hoan Luu Huu 96d72216e2 support google s2s host, version, sessionResumption (#1498) 2026-01-28 08:01:53 -05:00
Hoan Luu Huu faee30278b support mod_google_tts_streaming (#1409) 
* support mod_google_tts_streaming

* wip

* wip
 2026-01-27 08:18:47 -05:00
Hoan Luu Huu 325af42946 speechmatics support end_of_utterance_silence_trigger (#1499) 
* speechmatics support end_of_utterance_silence_trigger

* wip
 2026-01-23 10:11:58 -05:00
Hoan Luu Huu 9848152d5b support google gemini tts (#1491) 
* support google gemini tts

* wip

* wip

* wip

* wip

* wip

* wip

* wip

* update speech utils version

* wip
 2026-01-22 10:12:06 -05:00
Sam Machin 2468557aef add statusHook to redirect verb (#1500) 
* add statusHook to redirect

* fix url import

* Update redirect.js

* logging

* constructor statusHook

* lint n logging

* debug

* update call_status_hook

* use notifier to test url

* remove require url as its global since node 10

* update verb specs dep

* update verb specs
 2026-01-21 09:20:47 -05:00
Dave Horton 3c3dfa81d3 fix issue where call hangs up and actionhook delay triggered (#1497) 2026-01-19 16:42:43 -05:00
Vinod Dharashive 961c2589ac freeswitch capture sip error and propagate the same error (#1489) 
* fix: propagate SIP 488 error to SBC on endpoint allocation failure

When FreeSWITCH returns a SIP 488 'Not Acceptable Here' error during
endpoint allocation (e.g., codec incompatibility), this error was not
being propagated back to the SBC/client. Instead, the call would wait
indefinitely for websocket commands or return a generic 603 response.

Implementation:
- In _evalEndpointPrecondition(), detect SipError by checking
  err.type === 'SipError' or err.name === 'SipError'
- Extract the SIP status code (e.g., 488), reason, and the Reason
  header from the error response (e.g., Q.850;cause=88;text=INCOMPATIBLE_DESTINATION)
- Send the SIP error response immediately to the SBC with:
  - X-Reason header: endpoint allocation failure details
  - Reason header: original Q.850 cause from FreeSWITCH
- Notify call status change as Failed with proper SIP status
- Release the call immediately instead of waiting for commands

Also added fallback handling in InboundCallSession._onTasksDone() to
propagate the stored error if immediate send was not possible.

* wip

* Simplify SipError check to only use err.name
v0.9.5-10 		2026-01-13 08:58:37 -05:00
Hoan Luu Huu e4ec0025c3 Fix/gladia multi sessions (#1487) 
* support gladia transcribe multi channels

* wip
v0.9.5-9 		2026-01-07 10:46:33 -05:00
Ed Robbins ba275ef547 #1485 remove deprecated call to URL.parse (#1488) 2026-01-05 15:53:32 -05:00
Dave Horton 83a8cf6d25 SIGUSR1 should cause fs to commence drying up calls but do not exit when call count reaches zero (#1486) 2026-01-04 12:23:26 -05:00
Sam Machin 09220872ae Update recording (#1483) 
* refactor recording

removed the test of `(this.cs.accountInfo.account.record_all_calls || this.cs.application.record_all_calls` from backround-task-manager.jsL138 as this check is already done in call-session.js at Line 3007, also allows us to start the record from update or config verbs

* handle start recording for a call that is not yet answered

* return false if not changing recording state

* different check for status

* set hasRecording flag on callInfo when starting

* update redis on recording start

* lint

* update dependency
 2026-01-02 11:05:38 -05:00
Dave Horton fdce05fa40 add handler for SIGUSR1 to start drying up calls, useful as a generic mechanism on non-AWS deployments (#1482) 2025-12-30 13:31:42 -05:00
Sam Machin 3bd1dd6323 put removeListner in a try/catch (#1479) 
* put removeListner in a try/catch

* typo
 2025-12-19 13:31:06 -05:00
Ed Robbins 54dc172ebd Allow defining an ENV for specific webhook error return SIP code (#1476) 2025-12-16 17:14:42 -05:00
Hoan Luu Huu e007e0e2d3 fixed callsession cannot close tts streaming (#1472) 2025-12-16 07:58:54 -05:00
Hoan Luu Huu c5cd488fdf fixed gather should ignore transcription if task is killed/resolved. (#1465) 
* fixed gather should ignore transcription if task is killed/resolved.

* wip
v0.9.5-8 		2025-12-12 09:03:08 -05:00
Sam Machin 57982335e0 add label to STT/TTS alerts (#1468) 
* add label to STT/TTS alerts

* update time-series
 2025-12-11 11:07:24 -05:00
Hoan Luu Huu 5cea91e18a add support for sending DTMF to ultravox (#1471) 2025-12-11 07:53:59 -05:00
Dave Horton e396b6aa98 fix #1466: (#1467) 
* fix #1466:

* do not send tts streaming events when we are not doing tts streaming
v0.9.5-7 		2025-12-09 09:43:53 -05:00