Most often we have the problem of the VoIP customer not being able to hear the caller. This points towards packet loss during call setup or the customer's router not sending through the audio traffic back to the phone.
Enable TLS and SRTP on all extensions.
Temporarily enable call recording on their server. (Check with the customer if this is okay first). Explain to the customer this is so we can ascertain by listening to the recording where the fault lies. We can delete the recordings when the investigation is over.
Ask them for at least 2 further examples. Listen to the recording of these examples and ascertain that the server can hear both sides of the conversation.
If both sides can be heard on the recording then set up separate port ranges for incoming audio on each phone. E.g Phone 1 - 1000 to 1009, Phone 2 1010 - 1019 etc. You can find the settings for RTP port range in Yealinks under Network → Advanced → Local RTP Port
Now port forward the ranges to each phone IP address in their router.
Finally give each phone a different Local SIP Port. Ensure this port does not conflict with any of the RTP port ranges.
If one side can't be heard on the recording, then double check this is not a one off by looking at the other examples you have been given. Then:
For not being able to hear the external caller, we need to send RTP traces to the provider for the number.
For not being able to hear the VoIP customer we need to start investigating whether the audio traffic is reaching us at all.
We have had a case of a T41P model phone ignoring the set Local RTP Port range. This phone would only use the 119xx range of ports. If you get no luck using port forwarding then ask VIP support to find out what RTP ports the phone is requesting. In the case of the T41P, we adjusted the port forwarding in the router to match the range of the phone's requests and the problem was solved.