@juha-h please use ``cherry-pick -x <SHA>`` when back-porting
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#### Type Of Change
- [ x] Bug fix (non-breaking change which fixes an issue)
- [ ] New feature (non-breaking change which adds new functionality)
- [ ] Breaking change (fix or feature that would change existing functionality)
#### Checklist:
- [x ] PR should be backported to stable branches
- [ x] Tested changes locally
- [ ] Related to issue #XXXX (replace XXXX with an open issue number)
#### Description
Add mandatory Max-Forwards header to local ACK
You can view, comment on, or merge this pull request online at:
https://github.com/kamailio/kamailio/pull/3856
-- Commit Summary --
* Add mandatory Max-Forwards header to local AC
-- File Changes --
M src/modules/tm/t_msgbuilder.c (5)
-- Patch Links --
https://github.com/kamailio/kamailio/pull/3856.patchhttps://github.com/kamailio/kamailio/pull/3856.diff
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#### Description
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When calling delete_urecord in DB_ONLY mode the location_attrs are not deleted. This fix makes sure to delete location_attrs when calling db_delete_urecord is called.
You can view, comment on, or merge this pull request online at:
https://github.com/kamailio/kamailio/pull/3827
-- Commit Summary --
* usrloc: delete location_attrs in db_only mode on delete_urecord
-- File Changes --
M src/modules/usrloc/ucontact.c (36)
M src/modules/usrloc/urecord.c (4)
-- Patch Links --
https://github.com/kamailio/kamailio/pull/3827.patchhttps://github.com/kamailio/kamailio/pull/3827.diff
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These below is sample diagram, where kamailio/rtpengine act as an SBC
<img width="1604" alt="image" src="https://github.com/kamailio/kamailio/assets/143789335/b9540fb3-c1bf-4a78-b1…">
* There is an active call carrier, kamailio/rtpngine and Asterisk1
* Asterisk1 went down (power shortage, or network ..)
* What I want is kamailio self-redirect the call to Asterisk2
**Describe the solution you'd like**
Ideally, kamailio provide the command line to tell which call will be self-redirect to the new destination, eg:
`kamcmd -x sefl_redirect_callid <call-id> <destionation-ip>`
Noted: I read the https://blog.opensips.org/2021/06/09/media-re-anchoring-using-opensips-3-2/ but re-INVITE to carrier seem not appropriate in my case due to call rate limit. I'm looking for a new solution where I don't need to use re-INVITE when Asterisk down.
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