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Correcting One-way Audio with a VoIP Call - Asterisk
https://www.asteriskpbxsystems.com/troubleshoot-oneway-audio.html
Troubleshooting One-way Audio in VoIP by taking these steps. Fixing one-way audio issues in VoIP are best done one step at a time.Start by eliminating any double NAT possibilities by disabling NAT on any secondary routers that may be …
MeetMe() - Asterisk: The Definitive Guide (3rd edition)
http://asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-B-109.html
Synopsis MeetMe([confno[,options[,PIN]]]) Places the caller in to the audio conference bridge specified by the confnoargument. If the conference number is omitted, the user will be prompted to enter one. If the PINargument is passed, the caller must enter that PIN number to successfully enter the conference.
Playback in meetme (asterisk) - Stack Overflow
https://stackoverflow.com/questions/7021912/playback-in-meetme-asterisk
One easy way to do this is to have your asterisk system dial in (join) the the conference and then play the audio file then hangup. The call file will look something like this: Channel: Local/1000@default Application: Playback Data: sound_file Just make that file with the appropriate values and the place it in the call directory.
One way audio problem in trunk - Asterisk SIP - Asterisk ...
https://community.asterisk.org/t/one-way-audio-problem-in-trunk/72567
Hello. I got some problem for Asterisk SIP trunk. It’s Asterisk 11.17.1. SIP trunk from an operator. Outgoing call : Everything is ok Incoming call : Call signaling is ok but audio RTP is one way. Peer A,B and C can send audio packet to Peer X but it isn’t able to receive audio packet from Peer X. Network : [Asterisk trunk] ------- [VoIP Provider's Router] ---------- [VoIP …
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