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RTP300 PSTN Device - Need some help | 3CX Forums
https://www.3cx.com/community/threads/rtp300-pstn-device-need-some-help.20892/
I've been tinkering with 3cx on my home server. I have Windows Server 2003 and a Linksys RTP300. Incoming calls are working fine for audio, (not faxes yet, working on that). My problem is outgoing calls. It will ring on the outside line, but when I pick up, there is no audio. I …
Linksys Official Support - Getting to Know the RTP300
https://www.linksys.com/us/support-article?articleNum=133307
No failure or delay in exercising any right or remedy shall operate as a waiver of any such (or any other) right or remedy. The language of this Agreement shall not be construed strictly for or against either party, regardless of who drafted such language or …
Instruction manual RTP 300 Bedienungsanleitung RTP 300
https://ratio-tec.com/pub/media/wysiwyg/Dateien/user_manual_RTP300.pdf
7 An-/Ausschalten: Verbinden Sie den RTP 300 über die mitgelieferten Datenübertragungskabel mit dem gewünschten Gerät. Schalten Sie den Belegdrucker durch kurzes Drucken der …
PJSIP no audio on calls - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/pjsip-no-audio-on-calls/85165
Everything worked perfectly on chansip. I moved to PJSIP and I can’t hear audio on any of my calls. NAT was nat=force_rport,comedia for extensions 200 and 300 and nat=no for 100. It worked well and I used the python script to convert sip.conf to pjsip.conf. If I enable direct media, I’m able to hear one-way. PJSIP config: removed RTP debug shows no log, nothing.
Transmitting Audio and Video using RTP - Oracle
https://www.oracle.com/java/technologies/javase/transmitting-audio-video-rtp.html
The sample program can be used as a utility class as well as standalone. There is a main method that takes three command line parameters. If all goes well, the media will be sent out for 300 seconds and then the application will exit. This 300 seconds limit is because there is no GUI that allows you to specify when to stop the transmission.
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