Dave Horton
ffa006225b
fix amd timers
2022-08-01 12:20:12 +01:00
Dave Horton
2882fa2d0a
Feature/vm detection ( #137 )
...
* initial changes for amd
* wip
* fix bug where transcripts were discarded
* a bit of refactoring, and adding support for avmd in config verb
* bug fixes
2022-07-27 17:46:52 +01:00
Dave Horton
a035b67e6c
bugfix: hold music fetched when conference member removed from hold
2022-07-27 11:37:12 +01:00
Dave Horton
6979affb86
Feature/fast http client ( #132 )
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* initial changes to use undici for http client and connection pooling
* use body.json() mixin
* logging
* add pipelining env var
* implement socket close
2022-07-18 15:32:03 +02:00
Dave Horton
bb9c3a8df0
createCall: return callId along with sid
2022-07-12 09:56:13 +02:00
Dave Horton
7f808c6107
listen: when passDtmf is true, send dtmf events down websocket connection as json test frames ( #129 )
2022-07-07 11:55:29 +02:00
Dave Horton
f95524863d
update parse-url, improve Dockerfile
2022-07-06 19:16:31 +02:00
Dave Horton
aceaa5b7da
bugfix: continuous asr - if ended by dtmf allow collection of final transcript
2022-06-28 10:10:31 -04:00
Dave Horton
7d57c85153
bugfix #121 : Dial verb not ending when call no answer timeout exceeded
2022-06-24 10:50:29 -04:00
Dave Horton
9aa0df256d
initial changes to support siprec recording ( #120 )
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* initial changes to support siprec recording
* include additional params on SIP INFO to start recording
* add support for maniupulating recording via REST API
* fixes from testing pause/resume recording
2022-06-23 16:21:35 -04:00
Dave Horton
627c38899f
Feature/continuous asr ( #119 )
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* bugfix: background gather for speech-only should still kill audio on dtmf entry when dtmfBargein is true
* initial changes for continuous asr
* move properties under recognizer
* update drachtio-srf@4.5.1
* catch exception on destroy
2022-06-21 10:35:27 -04:00
Dave Horton
12ad7e556f
added support for sip:request verb, used to send SIP INFO/NOTIFY etc during call ( #116 )
2022-06-15 13:31:32 -04:00
Dave Horton
05d6c8d467
linting
2022-06-14 08:24:44 -04:00
akirilyuk
5e9407ff4e
add defaults to rest call payload ( #115 )
...
Co-authored-by: akirilyuk <a.kirilyuk@cognigy.com >
2022-06-14 08:20:55 -04:00
Dave Horton
5be5b6d05d
bugfix: broken enqueue waitHook ( #113 )
2022-06-11 10:38:35 -04:00
Prashanth
7f0e373e5f
issue# 107: on gather timeout, if minDigits are collected, resolve wi… ( #111 )
...
* issue# 107: on gather timeout, if minDigits are collected, resolve with dtmf-num-digits
* gather timeout: use conditional instead of if/else
Co-authored-by: Prashanth Gujjeti <prashanth@minervacq.com >
2022-05-17 12:53:10 -04:00
Dave Horton
c3e5ffa52d
bugfix: transcribe of a dialed call can now occur on both legs
2022-05-15 13:45:55 -04:00
Dave Horton
6d34850dc6
bugfix: transcribe Azure interim transcripts were missing
2022-05-11 19:22:14 -04:00
Dave Horton
76ff1835a6
background gather listen only once for vad and other interrupt events
2022-05-11 09:21:54 -04:00
Dave Horton
a4e358596e
emit vad event on partial transcript
2022-05-10 15:14:10 -04:00
Dave Horton
c412554c6b
WsRequestor: reconnect if socket dropped from far end
2022-05-09 12:14:13 -04:00
Dave Horton
34fe22f6e1
minor
2022-05-08 16:34:42 -04:00
Dave Horton
182ad8c716
expose model and singleUtterance to gather/transcribe when using google
2022-05-08 12:29:55 -04:00
Dave Horton
036accab44
dial: transcribe and listen should be based on the caller (A leg) endpoint
2022-05-07 18:36:49 -04:00
Dave Horton
b37881a059
bugfix: second part of outbound dial fix over wss
2022-05-07 11:52:29 -04:00
Dave Horton
258e4b5434
bugfix: outbound rest dial over websocket api needs to send session:new
2022-05-07 11:51:21 -04:00
Dave Horton
aa4d72c80a
allow call status to be sent before killing rest dial on failure
2022-05-02 14:05:24 -04:00
Dave Horton
5c38ace5ba
bugfix: rest dial should exit upon call failure, not after call timeout is reached
2022-05-02 13:50:42 -04:00
Dave Horton
dea58c2605
more work on wss race condition
2022-05-02 13:32:07 -04:00
Dave Horton
eb0f55e0e3
ws-requestor: queue outgoing messages if we are in the process of connecting to the remote wss server
2022-05-02 13:09:23 -04:00
Dave Horton
daa02ac55a
logging
2022-05-02 11:12:39 -04:00
Dave Horton
5134d5dbc6
update to latest realtimedb-helpers
2022-05-02 10:55:42 -04:00
Dave Horton
a755e25568
minor logging
2022-05-02 10:21:17 -04:00
Dave Horton
13549286db
bugfix: createCall needs to work with wss url
2022-05-02 09:42:04 -04:00
Dave Horton
72aaf80335
add support for multiple languages when using Azure STT
2022-04-26 15:07:55 -04:00
Dave Horton
85d86cfdc3
bugfix: gather catch errors when webhook fails
2022-04-24 13:45:29 -04:00
Dave Horton
de9f2ce5ca
bugfix: handle error if we cannot get our own ipv4
2022-04-21 19:09:23 -04:00
Dave Horton
36c97e9562
simplify error message
2022-04-21 14:43:09 -04:00
Dave Horton
13ea559cb1
send error notification over websocket if tts fails
2022-04-21 14:33:49 -04:00
Dave Horton
698d12a95f
clean up error handling in say verb
2022-04-21 10:27:33 -04:00
Dave Horton
359cb82d80
per recommendation from microsoft, do NOT sort transcripts by confidence: first transcript in the returned list is 'best'
2022-04-17 17:53:16 -04:00
Dave Horton
29dec24095
bugfix: azure stt - if we get no speech detected, listen again
2022-04-13 12:07:30 -04:00
Dave Horton
24a0bc547f
gather: dont restart transcribing if task has been killed
2022-04-11 21:13:49 -04:00
Dave Horton
db5486de27
gather bugfix: dont start transcribing after call is gone
2022-04-10 15:48:35 -04:00
Dave Horton
41d6c74c8e
send application defaults for speech in initial webhook
2022-04-09 11:38:31 -04:00
Dave Horton
92ca40c9b3
add feature flag env JAMBONES_INJECT_CONTENT ( #98 )
2022-04-06 15:54:59 -04:00
Dave Horton
3fa913215f
bump version
2022-04-06 08:19:33 -04:00
Dave Horton
077d34dc9e
gather: resolve with reason killed prevented task from ending
2022-04-05 08:15:30 -04:00
Dave Horton
49a75a3e3a
minor logging improvement
2022-04-04 14:02:09 -04:00
Dave Horton
6f214a66e8
AdultingSession: just create new child logger (simpler)
2022-04-04 13:49:30 -04:00
