I just spun up the latest version of RasPBX, Asterisk on the Raspberry Pi. Asterisk is version 16.13.0, and FreePBX is 15.0.17.55.
There are four Grandstream VOIP phones as extensions, all registered as PJSIP, and there's one SIP trunk going out to the world. The PBX and the phones are on the same subnet. All of the phones can receive audio, but none can send audio to another device. 1. Call the speaking clock at *60, it works just fine, Allison's voice is heard. 2. Call from one extension to another, no audio. They're probably both receiving audio, but neither can send any. 3. Make a call outgoing on the SIP trunk, the local phone can hear the other party but cannot be heard. 4. Receive a call from the SIP trunk - again, the phones can hear the other party, but cannot be heard. I don't think it can be any kind of NAT issue, since the phone extensions and the PBX are on the same subnet. Aside from the phones not sending audio, everything seems to be working. If I bring up Asterisk Info in FreePBX it shows the extensions and the trunk using PJSIP with ONLINE status. It's got to be something simple and dolt-ish I'm doing or have failed to do. Any ideas? Thanks, Curt Lundgren -- -- You received this message because you are subscribed to the Google Groups "NLUG" group. To post to this group, send email to [email protected] To unsubscribe from this group, send email to [email protected] For more options, visit this group at http://groups.google.com/group/nlug-talk?hl=en --- You received this message because you are subscribed to the Google Groups "NLUG" group. To unsubscribe from this group and stop receiving emails from it, send an email to [email protected]. To view this discussion on the web visit https://groups.google.com/d/msgid/nlug-talk/CADPPtrofatmmG310FD%2Bo7ZsjP%2B-BRS%2BviQ3fdgTebpb_Akkcng%40mail.gmail.com.
