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PBX Sending SIP traffic to wrong port

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Dewan_Fourie

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Good day,

I am trying to set up a Yealink P60B as a remote extension using STUN. The phone has been auto provisioned, and it can take and make calls, but no sound is present when answering the phone. I've tracked SIP traffic through the PBX, and it seems like the PBX is contacting the correct remote public ip when receiving a call, but on a seemingly random port (3209 - see attached). My router does not have SIP ALG functionality, but just to make sure, I've forwarded all the necessary ports for RTP, SIP and TLS SIP. Please see recorded traffic attached.

Thank you in advance.
 

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Hello @Dewan_Fourie

Since the phone is using STUN the PBX probably sees the phone coming from that phone. With STUN phones the firewall in the location the phone is also comes into play. Did you assign a unique SIP port to the phone and port forward that on the firewall? https://www.3cx.com/3cxacademy/videos/intermediate/configuring-remote-extensions/
Also since calls work then the SIP port seems to work so the issue might be with the audio ports or codecs.
Call *777 from your extension and see if you have audio both ways. That will give us a hint as to where the issue is.
You may also need to run a capture on the PBX and analyze it to see where the issue originates.
 
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Hi YiannisH_3CX and JohnS_3CX,

Thank you so much for the information, it narrowed it down for me quite a bit. I will use the information you gave me and will report back on my findings.
 
I've just tested *777, and it worked. I assumed this meant that the voice traffic was working, and I retried a call to this extension, and all is working fine now. I didn't make any changes to any config, but it seems to be working now so I'll just take the 'w'.

Thank you for your help!
 
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No problem, but with STUN you might get periods of times where it works and then stops again. This may until the changes in the article above have been performed or unless you use an SBC that largely bypasses the need to make those changes.
 
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