Hello All, I am new to SIP/VOIP and 3CX but have found 3CX to great and very easy to learn and implement. I am running into a problem now that I cannot seem to pinpoint, whether it could be a simple setting change in the product or something that might require alterations made outside of the product. The problem is I continually (but not all the times) run into problems with others hearing my side of the conversation. In some cases this can be seen if someone calls my extension externally or if someone who has a extension registered to the system is outside of the office. They will call me and I can hear them but they cannot hear me. I have done the standard firewall checker in 3CX and it passes and going through the docs it appears all the ports are open properly. Below is the latest communication from the server when this happened. What happened is someone that is external to the office registered ext.4111 and then tries to call someone in the office at ext.1104. The call is placed,1104 rings, picks up, and can then hear 4111 speaking but 4111 cannot hear 1104. It happens also if 4111 hangs up and calls a cell phone or any other external number. 13:38:25.901 [CM503008]: Call(35): Call is terminated 13:37:52.533 Session 120166 of leg C:35.1 is confirmed 13:37:52.221 [CM503007]: Call(35): Device joined: sip:email@example.com:5062;transport=udp 13:37:52.221 [CM503007]: Call(35): Device joined: sip:firstname.lastname@example.org:50204 13:37:52.221 [MS210003] C:35.1:Answer provided. Connection(transcoding mode[unsecure]):188.8.131.52:9024(9025) 13:37:52.205 [MS210001] C:35.2:Answer received. RTP connection[unsecure]: 10.0.0.121:5004(5005) 13:37:52.205 Remote SDP is set for legC:35.2 13:37:49.663 Active calls counted toward license limit:  13:37:44.171 [CM505001]: Ext.1104: Device info: Device Identified: [Man: Grandstream;Mod: GXP Series;Rev: General] Capabilities:[no-reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [Grandstream GXP2000 184.108.40.206] PBX contact: [sip:email@example.com:5060] 13:37:44.171 [CM503002]: Call(35): Alerting sip:firstname.lastname@example.org:5062;transport=udp 13:37:44.078 [CM503025]: Call(35): Calling Ext:Ext.1104@[Dev:sip:email@example.com:5062;transport=udp] 13:37:44.078 [MS210002] C:35.2:Offer provided. Connection(transcoding mode): 10.0.0.41:7070(7071) 13:37:44.062 [CM503004]: Call(35): Route 1: Ext:Ext.1104@[Dev:sip:firstname.lastname@example.org:5062;transport=udp] 13:37:44.062 [CM503010]: Making route(s) to ""<sip:1104@> 13:37:44.062 [MS210000] C:35.1:Offer received. RTP connection: 10.122.41.111:62414(62415) 13:37:44.047 Remote SDP is set for legC:35.1 13:37:44.047 [CM505001]: Ext.4111: Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA941-5.1.8] PBX contact: [sip:email@example.com:5060] 13:37:44.047 [CM503001]: Call(35): Incoming call from Ext.4111 to ""<sip:1104@> 13:37:44.015 [CM500002]: Info on incoming INVITE: INVITE sip:1104@ SIP/2.0 Via: SIP/2.0/UDP 10.122.41.111:50204;branch=z9hG4bK-31d52ce4;received=220.127.116.11 Max-Forwards: 70 Contact: "."<sip:firstname.lastname@example.org:50204> To: ""<sip:1104@> From: "."<sip:4111@>;tag=1a8795161471bd63o3 Call-ID: email@example.com CSeq: 102 INVITE Expires: 240 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Proxy-Authorization: Digest username="4111",realm="3CXPhoneSystem",nonce="414d535c0285e5e789:bddf3feeebc53b73249607881119a1e2",uri="sip:1104@......",algorithm=MD5,response="c62a6e7b62549da457b575450a3f5cea" Supported: replaces User-Agent: Linksys/SPA941-5.1.8 Content-Length: 0 Unfortunately I am not very proficient in reading the logs yet so looking at this I really don't see any clear explanation why this might be happening. If anyone has ever seen such a scenario and might have an idea on what to do please let me know. Thanks.