So i have stores that have two yealink t23g desk phones and then a w56p with two w56h cordless phones. For some reason when someone calls from an external number the caller can hear the store talking but the store cant hear the caller. If the store were to call them back then they would be able to hear each other. This is only happening on the cordless phones, desk phones work fine. If you were to call the store from an internal extension then the call would be fine. I’ve tried updating the firmware on the cordless phones, completely removing anything tied to that store in the freepbx phone system and recreating it, formatting the cordless phones/base. I’m not sure what else to try and any insight would be appreciated.
Sounds like the NAT configuration for your cordless phones isn’t working. How are they configured? Are they extensions configured in a central FreePBX server or are they connected through some other mechanism?
so we have our freepbx server cloud hosted by cyberlynk and then we used a dns name to connect the phones to the server
OK, so everything is behind firewalls and routers (your phones on one side and the VPS on another). Since the traffic works once you’ve opened a path (the outbound call works) but doesn’t when the call is coming in cold (there is no audio stream established to and from the phone and server), it looks to me like your either your router configuration is not working correctly, or the NAT settings in the extension (in the server) or in the phone itself (in the phones configuration) isn’t right.
Another possibility is that you’re forcing all inbound traffic at the “phone” end of the network to the wrong endpoint.
In fact, the more I think about it, the more I don’t know about how you are routing your traffic and getting things to work. Clearly, the “reverse” channel for the audio is broken, which usually indicates some kind of problem on the handset, but there are at least four devices that could be screwing you up.
Be sure to turn off all VOIP Helpers in any of your routers (SIP-ALG, for example, is a killer for Asterisk-based systems).
Without logs and/or SIP debug output, everything we could suggest is going to boil back down to “go back and read about one-way audio.” Without specifics, that’s probably the best anyone can suggest.
thanks for the advice, what kind of logs would you like me to pull for you to better help you understand what i have going on?
You’re probably going to need a SIP DEBUG trace for a phone call where the audio fails. Pay attention to the IP addresses in the SIP packets - that should give you some indication of where the problem is.
You can also look at the /var/log/asterisk/full server log to see what it thinks. While you’re searching, remember that (unlike real phones) when you calls are routed, they are always routed through the server. There’s no such thing as a peer-to-peer call in this scenario, so the interaction that’s failing is either from the server to your local extension or from the server to the ‘remote’ phone. Because of this, the audio could be failing between the server and either end of the conversation (in other words, you’ll need to pay attention to the incoming connection from your extension and the outgoing connection to the remote phone).
All it takes is one missed setting on any of the legs of this thing to screw up your audio.
did you receive my personal message with the logs? is that what you were asking about?
This topic was automatically closed 86 days after the last reply. New replies are no longer allowed.