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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...
Global variables like $${storage_dir} not recognized in param moh-sound in conference profile. Conference profile configuration: It cause error like: [WARNING] mod_sndfile.c:281 Error Opening File [/usr/local/freeswitch/sounds/en/us/callie/$${storage_dir}/conference/30017012/moh-sound.wav] [System error : No such...
FreeSWITCH Version:1.10.2~64bit ( 64bit) OS:Windos10/Centos7.6 I used a SIP phone to call the SIP.js webpage. After repeated tests, it was confirmed that the voice could not be transmitted before 2-5s...
for issue https://github.com/signalwire/freeswitch/issues/1698 after we get channel state at line 1995 in switch_ivr_bridge.c, it may be modified by other thread, so line 1956 "if judgement" maybe invalid.
For more details on Vosk see https://github.com/alphacep/vosk-server https://github.com/alphacep/vosk-api
Found what looks like a bug in the setting of session->read_frame_count. In switch_core_io.c (line 151) this assignment is used `session->read_frame_count = (session->read_impl.samples_per_second / session->read_impl.samples_per_packet) * session->track_duration; ` In switch_core_session.c (line...
Hello, **Describe the bug** FreeSwitch install by source compile on Centos7.9, then run the program package of freeswitch, but crash when to voice call. **To Reproduce** Steps to reproduce the...
This problem has shown itself on WebRTC. WebRTC is moving to hide local IP address with mDNS hostnames. For example: ``` a=candidate:2492242602 1 udp 2113937151 b8c27db8-0cb1-4252-ba37-dc32d0ac1dce.local 55693 typ host generation...
## run soundtouch on freeswitch 1.10.1 and Debian 10 ``` ``` ## it has no error, but can not receive any sound
Freeswitch version: FreeSWITCH Version 1.10.2-release.4~64bit (-release.4 64bit) on CentOS 7 Not sure this is a bug, but when enabling record_session for a bridged call, there seems to be something unusual...
The problem I wrote [here](https://github.com/signalwire/freeswitch/issues/920#issuecomment-744349000) is not fixed in version 1.10.6. Freeswitch does not include line "a=crypto:" in SDP if "rtp_secure_media_suites=AES_CM_256_HMAC_SHA1_80" (or AEAD_AES_256_GCM_8) is set in dialplan. **INVALID SUITE SUPPLIED**...