One way voice on Android Device

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alex98ti

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Hi, we have sucessssfully installed the 3cx V9 on our server and its working great. I downloaded Android client for Motorola Mileston and registered the account and able to make a phone call. The problem is I can not hear the other side from the android device but they hear me. I appreciate any help. I have ran firewall checks and tried other sip clients on this phone and iPhone and all same result. One way voice. Here is the log:

12:11:22.859 [CM503008]: Call(2): Call is terminated
12:11:06.609 Currently active calls - 1: [2]
12:10:43.765 [CM503007]: Call(2): Device joined: sip:[email protected]:5060
12:10:43.765 [CM503007]: Call(2): Device joined: sip:[email protected]:1733;transport=UDP
12:10:40.484 [CM505003]: Provider:[MyProvider] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:[email protected]:5060]
12:10:40.484 [CM503002]: Call(2): Alerting sip:[email protected]:5060
12:10:38.437 [CM503025]: Call(2): Calling VoIPline:14162222222@(Ln.10001@MyProvider)@[Dev:sip:[email protected]:5060]
12:10:38.390 [CM503004]: Call(2): Route 1: VoIPline:14162222222@(Ln.10001@MyProvider)@[Dev:sip:[email protected]:5060]
12:10:38.390 [CM503010]: Making route(s) to <sip:[email protected]>
12:10:38.390 [CM505001]: Ext.66: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone for Android 1.2.3] PBX contact: [sip:[email protected]:5060]
12:10:38.375 [CM503001]: Call(2): Incoming call from Ext.66 to <sip:[email protected]>
12:10:28.406 [CM504001]: Ext.66: new contact is registered. Contact(s): [sip:[email protected]:1733;transport=UDP/66,sip:[email protected]:5488/66]
12:10:28.171 [CM504002]: Ext.66: a contact is unregistered. Contact(s): [sip:[email protected]:5488/66]
12:10:27.500 [CM504001]: Ext.66: new contact is registered. Contact(s): [sip:[email protected]:42304/66,sip:[email protected]:5488/66]
 
Try enabling the STUN server settings.
 
Thank you for the response but I see STUN is enabled and I have an option to turn off STUN. Any advise is appreciated.
 
If you're using your local wifi network be sure to disable or re-configure your firewall to allow audio to pass through.
 
I am on 3G. Not sure how to set that on 3G
 
One way voice is almost always related to RTP Ports. Are the RTP Ports forwarded on the server end?

If you can get more detailed logs, check the Session Description section of the INVITE's.
 
Thanks for the reply. Would you please mention how RTP Ports is forwarded on the server? Also this log is the entire log I can get when I make a call and disconnect. I get two way audio on WiFi but not 3G. Same with iPhone.
 
My local RTP port becomes unreachable and shows an error "Destination unreachable (Port unreachable)" with mjsip...an advice is appreciated.
 
Check out Service Pack 5, there is a new feature and this solved my audio problems on T-Mobile.
 
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