drachtio drachtio-fsmrf issues

drachtio / drachtio-fsmrf

Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf

MIT License

48 stars 26 forks source link

issues

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startRecord erroring when starting conference

#51 lylepratt closed 2 years ago
0
Limit calls

#50 malvarado611 opened 3 years ago
0
what is sip_h_X-esl-outbound var ?

#49 herbiel closed 3 years ago
8
Bump y18n from 4.0.0 to 4.0.1

#48 dependabot[bot] closed 3 years ago
1
esl socket reconnecting repeatedly

#47 spointer closed 3 years ago
2
Problems overwriting User-Agent header

#46 spointer closed 4 years ago
2
SRTP Errors when used after failed a B2BUA with RTPEngine

#45 lylepratt closed 2 years ago
12
Adding support to override the X-esl-outbound header for NAT situations

#44 lylepratt closed 2 years ago
4
Override Private IP in X-esl-outbound

#43 lylepratt closed 2 years ago
1
Bump lodash from 4.17.15 to 4.17.19

#42 dependabot[bot] closed 4 years ago
1
Unable to connect Media Server ( Remote )

#41 amanullahtanweer closed 4 years ago
5
sending invite request to myself says error: "cannot call to myself"

#40 TareenAsad closed 4 years ago
5
Inbound calls from classic sip endpoint (RTP) to WebRTC endpoint (SRTP) not working.

#39 guy032 opened 4 years ago
17
Detected-Tone or Detected-Fax-Tone events not available

#38 guy032 closed 4 years ago
8
Add RECV_RTCP_MESSAGE to EVENTS_OF_INTEREST

#37 guy032 closed 4 years ago
1
Bump minimist from 1.2.0 to 1.2.3

#36 dependabot[bot] closed 4 years ago
0
Bump minimist from 1.2.0 to 1.2.2

#35 dependabot[bot] closed 4 years ago
1
Handling of error in playback

#34 WhiteyDude closed 4 years ago
1
unable to find dialog for dialog id provided

#33 WhiteyDude closed 4 years ago
8
any existing custom event callbacks are removed when endpoint joins conference

#32 davehorton closed 4 years ago
1
No playback on FreeSWITCH side after endpoint.play()

#31 enp closed 4 years ago
14
Unable to create endpoint

#30 wnieves19 closed 4 years ago
15
Recording and playing audio doesn't work as expected

#29 telmojsneves closed 4 years ago
2
Problems playing audio into call

#28 ruipfmendes opened 5 years ago
2
reconnection no longer working

#27 davehorton closed 5 years ago
1
Custom sofia profile branch

#26 byoungdale closed 5 years ago
6
conference events

#25 mpeguerus closed 5 years ago
6
Freeswitch with mulitple sip profiles

#24 amoonrah closed 5 years ago
17
Memory leak assistance

#23 WhiteyDude opened 5 years ago
3
Feature request: throw error when attempting to send a command to a non active endpoint

#22 WhiteyDude opened 5 years ago
0
Early hangup can lead to dead endpoint with continued execution

#21 WhiteyDude closed 5 years ago
9
connectCaller times out with webrtc client

#20 andrewvmail closed 4 years ago
52
callback variants of the api doesnt work

#19 andrewvmail closed 5 years ago
2
Handeling B leg destruction in bridge does not seem to work

#18 WhiteyDude closed 5 years ago
2
Some tests failed due to "Error: Error: connect ECONNREFUSED 127.0.0.1:9071"

#17 alishir opened 5 years ago
1
IPV6 NAT64 WebRTC network Freeswitch will error out

#16 andrewvmail closed 5 years ago
2
Usage of Bridge and Unbridge

#15 haeferer opened 5 years ago
2
Strange States on Endpoint CallStates with Bridge/Unbridge

#14 haeferer opened 5 years ago
3
Strange Behaviour using "tone_stream://%(1850,4150,475,425);loops=-1" on playback

#13 haeferer opened 5 years ago
4
using .say on endpoint works, but produces no audio

#12 haeferer opened 5 years ago
1
Removing an existing Conference

#11 haeferer closed 5 years ago
14
Correctly destroying Endpoint and Dialog

#10 haeferer opened 5 years ago
0
endPoint.play does not return if call end by RemoteParty

#9 haeferer opened 5 years ago
8
mediaserver.createEndpoint() does not return ...

#8 haeferer closed 5 years ago
2
Sample for Using with createUAC

#7 haeferer closed 5 years ago
2
msml support for drachtio

#6 ElliotGo closed 6 years ago
5
endpoint modify method does not send reINVITE to far end

#5 byoungdale closed 5 years ago
3
Video call between an intercom device (SIP) and a web client (WebRTC)

#4 jdzuri closed 5 years ago
4
Error: No drachtio_mrf sip profile found on the media server

#3 jdzuri closed 7 years ago
3
Troubles trying to record the user voice

#2 massimo-romano closed 7 years ago
5

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