freeswitch
https://github.com/signalwire/freeswitch
C
FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a Raspberry PI to a multi-core server, FreeSWITCH can unl
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- Issues
- -Werror=address -Werror=stringop-truncation
- Error Creating SIP UA for profile: internal (sip:[email protected]:5060;transport=udp,tcp)
- reinvite not sent to getway normally
- Freeswitch still negotiate SRTP even though rtp_secure_media set to forbidden
- How to get value of SIP header in Freeswitch v1.10.X?
- Fix signalwire/freeswitch#1965 where received audio payload is misinterpreted as DTMF
- play_and_detect_speech errors
- [mod_callcenter] Stale members listed as answered
- Added variables invite_contact_host and invite_via_host to change the host portion of the Via and Contact headers in INVITE requests
- Problem with rfc2833-pt and DTMF on bridged calls ever since.
- Docs
- C not yet supported