This sounds like an RTP stream issue. While SIP runs on port 5060 by default, the actual audio is run on much higher ports, 10000 and up. If your pbx system is not on a public address and you are forwarding traffic to your pbx via a router, check the RTP port range in your pbx and make sure you are forwarding those ports correctly. By putting the call on hold, you stop sending out audio traffic restart sending audio when you pick the call back up. By forwarding the entire RTP range to your pbx, this should fix your issue.
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