865068f158
add new CODEC_TRANSCODE option ( #225 )
...
* add new CODEC_TRANSCODE option
allows RTP engine to add additional codecs to the outgoing offer that wern't in the invite from Freeswitch so RTP engine will transcode outbound calls.
* update README for new env var descriptions
---------
Co-authored-by: Dave Horton <daveh@beachdognet.com >
2026-05-01 08:43:13 -04:00
4f65b4b585
support srtp for sips sipuri outbound call ( #224 )
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* support srtp for sips sipuri outbound call
* wip
* wip
* add env variable for disable the srtp for sipURI
2026-04-22 08:00:50 -04:00
d053b94e71
remove ice and dtls off ( #214 )
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* remove ice and dtls off if set in db
* lint
* more lint
2026-01-29 13:43:30 -05:00
b882c0de2d
Recording update ( #212 )
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* if call hasRecording then add url to cdr
* handle missing hasRecording value in redis
* typo
* lint
* use nullish coalescing for null response
* typo
2026-01-02 10:29:51 -05:00
b3fee43c7f
include codec-accept on answer to rtpengine during reinvites ( #209 )
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* include codec-accept on answer to rtpengine during reinvites
* attempt to simplify
* fixes from testing
2025-12-02 07:37:18 -05:00
5fab8a7515
when far end answers with only pcma, passthrough instead of transcoding to pcmu ( #208 )
2025-11-24 11:23:16 -06:00
70a09c10b3
Fix/200 ( #202 )
...
* Update call-session.js
* Update call-session.js
2025-10-28 16:58:54 -04:00
49bc11bbb6
update pino and eslint ( #201 )
2025-10-21 07:52:29 -04:00
cd0d360561
Redirect outbound user calls to private IP of other SBC ( #197 )
...
* redirect client calls to private address of other SBC
* remove unused util
* use address not port
2025-10-04 20:08:57 -04:00
32d82ed67d
Fix/193 ( #195 )
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* pass sip failure reason back to FS
* Update call-session.js
* update drachtio-srf dep
2025-09-12 09:21:13 -04:00
c17f27ab2c
fix prev commit ( #192 )
2025-09-04 07:57:38 -04:00
2fc570f731
when sending to retell with user starting with call_ dont prepend plus ( #191 )
2025-09-03 23:35:56 -04:00
c87f831868
Fix/transport in contact ( #190 )
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* use the req transport param in the From and Contact headers if set.
* Update call-session.js
2025-09-03 13:53:51 -04:00
616228bf09
update isPrivateVoipNetwork function ( #189 )
2025-09-02 08:02:59 -04:00
55fef10f0e
revert change (for now) that caused audio issues when reinviting to partial media ( #187 )
2025-08-18 12:45:48 -04:00
9357920f76
set strict source ( #185 )
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* set strict source
RTPBleed
* change to env var for strict source
* Update srtp-transcoding.json
* lint
* lint
* reverse the logic
* and argghhh
* clarification
* change
2025-08-03 19:44:31 -04:00
e422c2ed9c
increase dtmf volume ( #184 )
...
https://github.com/jambonz/jambonz-feature-server/issues/1272
2025-07-09 08:22:53 -04:00
fd5e2f1a6c
Remove video sdp incase of reinvite ( #183 )
...
https://github.com/jambonz/sbc-outbound/issues/182
2025-07-08 09:17:16 -04:00
3a467921a6
return 482 loop detected if call to a sip uri would loop back to us ( #173 )
...
* return 482 loop detected if call to a sip uri would loop back to us
* this pr fixes #172
* typo
* wip
* wip
2025-03-20 11:25:08 -04:00
28dae50202
REFER should have sips contact if far end is using sips ( #168 )
2025-02-24 09:47:47 -05:00
7380457b5a
reject calls on hosted jambonz with no activ ( #166 )
...
e subscriptions
2025-02-19 13:04:44 -05:00
9feb6f3c8f
support voip carrier sip proxy ( #165 )
...
* support voiip carrier sip proxy
* wip
* wip
* wip
2025-02-17 09:47:59 -05:00
c449feeb9c
support sip recording from siprec call ( #164 )
...
* support sip recording from siprec call
* update srsclient version
2025-02-12 09:24:33 -05:00
b2abe9891e
refined the method for syncing call count updates and added debugging ( #163 )
...
* refined the method for syncing call count updates and added debugging
* wip
* include callId in debug key
2025-02-05 13:01:04 -05:00
d8ae824559
Fresh/466 ( #160 )
...
* Hold and unhold does not resume transcript for outbound call to webrtc
* WIP
* jslint
* jslint
* handle null
* Added direction
* re-invite with opus codec need to change pcmu
on reinvite on unhold opus codec was been sent from rtpegine to freeswitch due to which transcripts were not getting generated , hence it needs to be changed to pcmu
* jslint
jslint
2024-12-23 07:20:04 -05:00
334db6f84f
support referby display name ( #161 )
2024-12-11 10:34:27 -05:00
c762accce8
forward all refer to feature server ( #156 )
...
* forward all refer to feature server
* forwawrd extra custom header
2024-12-05 21:16:12 -05:00
c6c63e26da
fix duplicate attempt to destroy dialog
2024-11-22 10:46:24 -05:00
916d577b75
wip ( #155 )
2024-11-19 09:37:53 -05:00
b976a62a60
when forcing PCMU or PCMA we must also include telephone-event ( #153 )
2024-11-14 08:37:40 -05:00
5e2369e5e8
support opus transcode ( #150 )
...
* support opus transcode
* dont resend reinvite when release media for srtp
* wip
* fix review comment
2024-11-13 07:41:43 -05:00
41528db630
fix call count race condition #151 ( #152 )
...
* fix call count race condition #151
* logging fix
2024-11-12 14:57:04 -05:00
c22104fbe2
allow outbound cdr has sip_parent_callid ( #149 )
...
* allow outbound cdr has sip_parent_callid
* update time series version
2024-10-17 08:05:25 -04:00
3701e20295
fix sbc crash while outbound calling to user ( #148 )
2024-10-15 07:28:40 -04:00
821275cb67
support X-CID for feature server to detect sip callid ( #147 )
...
* support X-CID for feature server to detect sip callid
* fixed review comment
* fix review comment
2024-10-15 07:19:25 -04:00
f5ac51a8f7
minor logging
2024-10-13 10:12:55 -04:00
0cf90f37b8
support change log level runtime ( #146 )
2024-10-07 09:52:22 -04:00
b8fdf9f429
when sending out registered trunk use auth.username as userinfo in Contact header of INVITE
2024-09-04 13:40:59 +01:00
503aa73bc6
Fix/invite use realm to registered trunk ( #144 )
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* special header X-Preferred-From-Host was ignored
* when sending INVITE to registered trunk with a sip realm, use realm in the uri and send via proxy as defined in outbound gateway
2024-08-29 15:26:44 -04:00
a7406ddb8a
use system_information.private_network_cidr ( #143 )
...
* use system_information.private_network_cidr
* make gh actions work
* fix: export logger
2024-08-18 12:50:19 -04:00
bf1c670b40
Fix/tls transport ( #141 )
...
* fix scheme
* add missing initialization of scheme
* delete contact header explicitly
* wip
* wip
* wip
* fix bug with cseq
* wip
* deps
* wip
* wip
---------
Co-authored-by: Markus Frindt <m.frindt@cognigy.com >
2024-07-24 15:27:01 -04:00
b21eaad1ff
Support contact header with sips or sip over tls ( #139 )
...
Co-authored-by: Markus Frindt <m.frindt@cognigy.com >
2024-07-23 08:53:38 -04:00
73e3779eb1
only send Sips if it's enabled in outbound gateway configuration ( #136 )
...
* only send Sips if it's enabled in outbound gateway configuration
* update license
2024-06-15 09:12:02 -04:00
9ad7dc76c7
fix for #133 ( #134 )
2024-05-17 07:21:19 -04:00
1baf7fa824
Feat/refer support conplex uri for refer-to refered-by header ( #131 )
...
* respond to reinvite request incase error
* support complex refer-to refer-by uri
* wip
2024-05-07 08:28:08 -04:00
16a4709b7d
check if sip gateway is in blacklist before sending outbound call ( #119 )
...
* check if sip gateway is in blacklist before sending outbound call
* wip
* wip
* wip
* add testcase for blacklist
* wip
2024-03-30 11:10:13 -04:00
01adb5cbf0
fix cannot clear siprec records ( #127 )
2024-03-27 07:50:50 -04:00
49ba872f51
remove asymmetric flag on offer to B party reinvite as port learning should still happen ( #128 )
2024-03-26 12:13:01 -04:00
591b8f2f8a
respond to reinvite request incase error ( #126 )
2024-03-07 07:49:05 -05:00
16ac56e50e
start/stop/pause/resume recording success when one of siprec server success. ( #125 )
...
* start/stop/pause/resume recording success when one of siprec server success.
* wip
* allow send custom header on pause, resume recording
* allow timeout for siprec action
* wip
* wip
* update siprec-client-utils
2024-03-05 13:06:04 -05:00
Sam Machin