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### Feature Description
At work we need to listen to a 1400Hz special information tone for conference call detection as specified in [Swedish tones in public communications network](https://its.se/wp…
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I'm sure I just havent properly configured asterisk but for some reason I am not able to call any of the API's.
Testing the examples on the documentation all return 404. The resources.json url retu…
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**Describe the bug**
In outgoing calls I need to listen to the early media audio.
After configuring early media using the UserAgent api, it always returns a 403.
**Logs**
https://gist.github.com…
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Hi, is it possible to install this soft phone on top of freepbx and use it with freepbx existing user accounts and extensions?
Is there a guide on how to install it on top of freepbx?
Thank yo…
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Hello Team,
I am so happy to see the demo for go-auto dial on the wiki page. It's really good and smooth interface and looks like APIs will also be released.
Any specific dates when the codebase…
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rtp_handler method is not being called for rtp packets. i also turn on rtp packets debug on asterisk and don't see any rtp packets, call just not answering. log from asterisk:
`Using SIP RTP CoS ma…
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```
What steps will reproduce the problem?
1. Connect to manager
backup-mirror datacard-read-only # telnet 127.0.0.1 5038
Trying 127.0.0.1...
Connected to 127.0.0.1.
Escape character is '^]'.
Asterisk…
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### Severity
Trivial
### Versions
18.23.1
### Components/Modules
MixMonitor
### Operating Environment
raspbx 5
Debian GNU/Linux 12 (bookworm),
Freepbx 16
### Frequency of O…
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cd /etc/asterisk/
vi sip.conf
[general]
register=>1010:supersecret@sip.flagonc.com:5600/9999
[siptrunk]
type=peer
defaultuser=1010
remotesecret=supersecret
port=5600
insecure=invite
host=sip.flagonc.…
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**Describe the bug**
We have followed the process [mod_unimrcp](https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Modules/mod_unimrcp_6586728/) provisioning/installation process. Despi…