-
```
What steps will reproduce the problem?
1. set-up - 2 sipml clients webrtc2sip sip server (freeswitch)
2. send an IM from 1002 to 1003 after registration
What is the expected output? What do yo…
-
1:freeswitch webrtc webSocket disconnects and reconnects successfully, and the phone hangs up automatically after one minute;
2:freeswitch sends bye first, webrtc will delay receiving bye message f…
-
freeswitch rel 1.10.2
I am trying to establish a media session with a remote side using the AMR codec :
Remote side sends Invite with:
a=rtpmap:96 AMR/8000
bb1a871a-328b-4f76-8057-ef3906f1886f …
-
大神好!我按照您的文档一步步配置,TTS可以正常运行,听到声音,但是ASR 科大讯飞服务返回的识别结果是Completion-Cause: 002 no-input-timeout. 在UniMRCP服务器侧抓包发现Freeswitch发给UniMRCP服务器的RTP包的Payload内容全为7f7f7f7f7f7f7f7f..., 想请教您可能是什么原因造成的,万分感谢!
192.168.20…
-
```
What steps will reproduce the problem?
1. Register with a Freeswitch server.
2. Setup Freeswitch so when it receives an incoming call multiple phones
will ring, including sipdroid.
3. Place incom…
-
```
What steps will reproduce the problem?
1. Register with a Freeswitch server.
2. Setup Freeswitch so when it receives an incoming call multiple phones
will ring, including sipdroid.
3. Place incom…
-
```
What steps will reproduce the problem?
1. Register with a Freeswitch server.
2. Setup Freeswitch so when it receives an incoming call multiple phones
will ring, including sipdroid.
3. Place incom…
-
```
What steps will reproduce the problem?
1. Register with a Freeswitch server.
2. Setup Freeswitch so when it receives an incoming call multiple phones
will ring, including sipdroid.
3. Place incom…
-
```
What steps will reproduce the problem?
1. Upgrade to 1.4.5
2. Use multiple register on FreeSWITCH (two phones on one extension - one
standalone THOMSON 2030 and Sipdroid on G1).
3. Call this exte…
-
```
What steps will reproduce the problem?
1. Upgrade to 1.4.5
2. Use multiple register on FreeSWITCH (two phones on one extension - one
standalone THOMSON 2030 and Sipdroid on G1).
3. Call this exte…