- Joined
- Jul 18, 2018
- Messages
- 12
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Hi all,
Preface: I have almost no experience with remote phone setups an am a total noob.
I have a few remote employees who have Yealink phones configured to work over STUN/Direct SIP (connecting to public static IP where the PBX is). All of the phones and extensions are configured for sRTP and deliver SIP transport with TLS.
Everything generally works well, but the remote phones will not terminate calls. It takes the other end of the call to terminate the call. The same thing happens when the remote phones leave voicemails to other extensions -- they are not terminating.
Under the extensions, I have ensured that the remote phones have different local RTP audio ports, but i'm not sure if i am missing something else.
Does anyone have any experience or recommendations for me?
Additionally, should the remote phones still need to leverage 5060 even if all traffic should be over 5061 and the other RTP ports?
Preface: I have almost no experience with remote phone setups an am a total noob.
I have a few remote employees who have Yealink phones configured to work over STUN/Direct SIP (connecting to public static IP where the PBX is). All of the phones and extensions are configured for sRTP and deliver SIP transport with TLS.
Everything generally works well, but the remote phones will not terminate calls. It takes the other end of the call to terminate the call. The same thing happens when the remote phones leave voicemails to other extensions -- they are not terminating.
Under the extensions, I have ensured that the remote phones have different local RTP audio ports, but i'm not sure if i am missing something else.
Does anyone have any experience or recommendations for me?
Additionally, should the remote phones still need to leverage 5060 even if all traffic should be over 5061 and the other RTP ports?