Dave Horton
b3fee43c7f
include codec-accept on answer to rtpengine during reinvites ( #209 )
...
* include codec-accept on answer to rtpengine during reinvites
* attempt to simplify
* fixes from testing
2025-12-02 07:37:18 -05:00
Dave Horton
5fab8a7515
when far end answers with only pcma, passthrough instead of transcoding to pcmu ( #208 )
2025-11-24 11:23:16 -06:00
Anton Voylenko
178105acd7
chore: bump node version ( #205 )
2025-11-04 18:02:31 -05:00
Sam Machin
70a09c10b3
Fix/200 ( #202 )
...
* Update call-session.js
* Update call-session.js
2025-10-28 16:58:54 -04:00
Dave Horton
49bc11bbb6
update pino and eslint ( #201 )
2025-10-21 07:52:29 -04:00
Sam Machin
2ebe4d3d78
bump db_helpers for cache change ( #199 )
2025-10-15 11:40:12 -04:00
Sam Machin
cd0d360561
Redirect outbound user calls to private IP of other SBC ( #197 )
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* redirect client calls to private address of other SBC
* remove unused util
* use address not port
2025-10-04 20:08:57 -04:00
Sam Machin
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
Dave Horton
c17f27ab2c
fix prev commit ( #192 )
2025-09-04 07:57:38 -04:00
Dave Horton
2fc570f731
when sending to retell with user starting with call_ dont prepend plus ( #191 )
2025-09-03 23:35:56 -04:00
Sam Machin
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
Sam Machin
616228bf09
update isPrivateVoipNetwork function ( #189 )
2025-09-02 08:02:59 -04:00
Dave Horton
55fef10f0e
revert change (for now) that caused audio issues when reinviting to partial media ( #187 )
2025-08-18 12:45:48 -04:00
Sam Machin
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
Dave Horton
6771cbdefc
bump version
2025-07-15 11:46:40 -04:00
Vinod Dharashive
e422c2ed9c
increase dtmf volume ( #184 )
...
https://github.com/jambonz/jambonz-feature-server/issues/1272
2025-07-09 08:22:53 -04:00
rammohan-y
fd5e2f1a6c
Remove video sdp incase of reinvite ( #183 )
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https://github.com/jambonz/sbc-outbound/issues/182
2025-07-08 09:17:16 -04:00
sathish kumar pasham
5cc37265a9
Fix vulnerabilities by adding @babel/helpers@7.26.10 to dependencies ( #181 )
2025-06-27 08:04:03 -04:00
Dave Horton
1a9c438321
update version
2025-05-14 15:42:31 -04:00
Dave Horton
fb62fd9bf4
update db-helpers
2025-05-13 10:36:08 -04:00
rammohan-y
e02e0fa8c1
Updated drachtio version to 5.0.5 ( #177 )
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https://github.com/drachtio/drachtio-server/issues/424
2025-04-22 11:05:45 -04:00
Hoan Luu Huu
dd198105e3
update drachtio srf 5.0.4 ( #176 )
2025-04-18 07:50:21 -04:00
rammohan-y
a34491a356
Updated db-helper to 0.9.11 ( #175 )
...
https://github.com/jambonz/jambonz-feature-server/issues/1151
2025-04-14 08:22:09 -04:00
Dave Horton
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
rammohan-y
9cb75a0b6b
updated realtimedb-helper to 0.8.13 ( #169 )
2025-03-10 09:53:49 -04:00
Dave Horton
28dae50202
REFER should have sips contact if far end is using sips ( #168 )
2025-02-24 09:47:47 -05:00
Dave Horton
0b45e766b9
update drachtio-srf
2025-02-20 10:16:59 -05:00
Dave Horton
7380457b5a
reject calls on hosted jambonz with no activ ( #166 )
...
e subscriptions
2025-02-19 13:04:44 -05:00
Hoan Luu Huu
9feb6f3c8f
support voip carrier sip proxy ( #165 )
...
* support voiip carrier sip proxy
* wip
* wip
* wip
2025-02-17 09:47:59 -05:00
Hoan Luu Huu
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
Dave Horton
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
Dave Horton
a89b54b19e
update deps
2025-01-14 10:50:08 -05:00
Dave Horton
ab0f3ed2b5
update dractio-srf and mrf to main branch releases ( #162 )
2025-01-14 10:06:27 -05:00
Vinod Dharashive
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
Hoan Luu Huu
334db6f84f
support referby display name ( #161 )
2024-12-11 10:34:27 -05:00
Hoan Luu Huu
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
Dave Horton
c6c63e26da
fix duplicate attempt to destroy dialog
2024-11-22 10:46:24 -05:00
Dave Horton
916d577b75
wip ( #155 )
2024-11-19 09:37:53 -05:00
Dave Horton
b976a62a60
when forcing PCMU or PCMA we must also include telephone-event ( #153 )
2024-11-14 08:37:40 -05:00
Hoan Luu Huu
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
Dave Horton
41528db630
fix call count race condition #151 ( #152 )
...
* fix call count race condition #151
* logging fix
2024-11-12 14:57:04 -05:00
Hoan Luu Huu
c22104fbe2
allow outbound cdr has sip_parent_callid ( #149 )
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* allow outbound cdr has sip_parent_callid
* update time series version
2024-10-17 08:05:25 -04:00
Hoan Luu Huu
3701e20295
fix sbc crash while outbound calling to user ( #148 )
2024-10-15 07:28:40 -04:00
Hoan Luu Huu
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
Dave Horton
f5ac51a8f7
minor logging
2024-10-13 10:12:55 -04:00
Hoan Luu Huu
0cf90f37b8
support change log level runtime ( #146 )
2024-10-07 09:52:22 -04:00
Dave Horton
62fb716236
bump version
2024-09-04 13:42:02 +01:00
Dave Horton
b8fdf9f429
when sending out registered trunk use auth.username as userinfo in Contact header of INVITE
2024-09-04 13:40:59 +01:00
Dave Horton
503aa73bc6
Fix/invite use realm to registered trunk ( #144 )
...
* 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
Dave Horton
a7406ddb8a
use system_information.private_network_cidr ( #143 )
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* use system_information.private_network_cidr
* make gh actions work
* fix: export logger
2024-08-18 12:50:19 -04:00
