One way audio on incoming calls. The caller cannot hear us


System has been working for over a year. Not sure what changed but now, on most incoming calls the caller cannot hear us when we answer the phone. Some calls get through. Outgoing calls audio is good both ways.

Ticket #: 20200630161824
Status: Awaiting for GS Response
Product: UCM-62xx
Platform: Grandstream UCM
Created On: 30 Jun 16:18
Last Updated: 3 days ago
Title: No audio on incoming calls
Issue Type: Software
Description: About 20 GXP2160 on the firmware.

Grandstream has been silent despite my repeated requests so I am looking at the community for a little help. I would appreciate if some one can tell me what I should be checking.



port forwarding in place? sip alg disabled in router?


Thanks for helping.
the answer is Yes and Yes


Have the local LAN stated in the NAT page of SIP Settings? Have the external host or IP stated?


Your service provider should be able to assist with troubleshooting - have you opened a ticket with them yet? Since it is your outbound stream that is affected on incoming calls, not the inbound stream, the NAT description in your PBX sip.conf is not the problem. My guess is that either your firewall is blocking the RTP stream UDP traffic outright after it leaves your PBX - its destination IP address, or its target port, or both - or it is intercepting and rewriting the call Invite packet that sets up the call, causing your PBX to direct its RTP to an invalid IP. If you have the ability to make a packet capture in your PBX simultaneous to one made in the provider’s server, you should be able to determine whether to blame your PBX, your provider, or your firewall. My money is on the firewall.


Additionally, set NAT in the trunk settings to No./Disable.


External host xx,xxx,xxx,xx:5060
SDP: Checked
Ext UDP Port: 5060
Ext TCP Port: 5060
Ext TLS Port: 5061
Local Net address:
NAT in trunk settings is NOT checked

I opened a ticket with Flowroute, the VoIP provider. i have attached a screen shot of my router’s opened ports.


change NAT to 1:1 and see.


Perhaps something I should have mentioned earlier that i didn’t think was relevant. My ISP is Spectrum who has provided a modem. The modem is in bridge mode. On the house side we have a Adtran NetVanda 3140 router with three ports, one input WAN from the modem and two outputs LAN, one to the local network ( where the network server is the DHCP server to the PCs) and one to the phone system ( where the router is the DHCP server to the phones). The idea was to separate the two traffics and it has worked well over the years.
Do you see a problem with this setup and do you think i should proceed with what you are suggesting?


problem i see on the router, unless im mistaking the nat translation…it allows all external voip to inside ip range …

I would normally assign the sip providers external ip to the inside ip of the ucm

is that what is programmed ?


forgive my ignorance. where exactly should the sip provider’s ip be entered?


I am unsure what or how to assign things inside your router, but in a Mikrotik… I have external IP of SIP Provider --> destination NAT and point to the internal IP of the UCM in this case and I would assign the ports - UDP 5060 external to UDP 5060 internal and UDP 10000 to 20000 external to inside UDP 10000 to 20000 internal

Possibly could be in the following…picture…



I guess the question is that with 1 Wan port is there only 1 public IP or are there more and using aliases? How are you sorting the traffic so that SIP & RTP get to the correct LAN side? I assume the destination NAT is how instead of the 1:1 I suggested.

Take a capture of an incoming call and post.

If it was all working well before, then what changed?

What Kev is suggesting is to filter the input rules such that only your provider or others of need can access the inside. This would normally be done by specifying the source address just as you specified the destination address (UCM). Possibly found in firewall/acls.

Who is the ITSP?


thanks for the suggestion. i will see where to make the change as the NetVarta is not an easily configurable router, at least to me.


is only one static public ip address.

i took a capture of a problem call (Network Troubleshooting-Ethernet Capture) and it produced a .tgz file. Tried to upload to this reply but apparently .tgz files are not allowed. any suggestions?

the SIP provider (Flowroute), changed their ip addresses PoP (Point of Presence). since then, there is nothing but trouble.

looking where in my router i can make the suggested changes

the ISP is Spectrum


reading another post perhaps this is a way to share the capture


Please provide the scenario of the capture what is the expectation?


I call the office from outside (incoming call). the office picks up, they can hear me but i can’t hear them.


What number did you use to call into the office?




There is 2-way audio. You dialed in, the phone was answered, but it seemed as though you apparently did not hear it. It was hard to say as you kept talking, but the lady on the other end only asked once if you could hear her, then she kept quite while listening to you.

The only thing that comes to mind is if the Adtran is re-writing source ports. The provider may be able to detect this, but to be on the safe side, both they and you need to do a capture of the same call at the same time.