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No outgoing audio on Outbound-Inbound calls

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Triantafillos Paradas

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Hello,
I have the following problem and I would like to hear your opinion.

1. When I make an outbound call, using the sip trunk, I can hear the other end but they can not hear me.
2. The same applies to inbound calls. I can hear them but they don't.
3. When I call another extension number, in any network combination LAN<->LAN, WAN<->LAN or WAN<->WAN I have no problem.

My system is a virtual machine with Debian installation v15.5.13103.5 with 2 NICs, one for the company's LAN/internet and the second goes directly to the VOIP provider's (OTE) router (no gateway defined at this interface). The firewall test passes successfully.

Is there something I can do to fix it?

Thank you in advance.
 
Hi Triantafillos,

One way audio primarily is a firewall issue (the correct ports need to be configured for pass-through):
https://www.3cx.com/ports-used-3cx-phone-system-v14-v15/

Most firewalls allow outbound traffic by default but block inbound.
https://www.3cx.com/support/firewall-configuration/

Also ensure that you have SIP ALG disabled on the Firewalls as well.

Also check your settings on the Trunk, is it a supported trunk ? do you have PBX delivers audio configured on the trunk itself ?
 
Hello eddv123,
My firewall ports, I believe, they are OK:
Ports.jpg

My firewall inbound traffic is OK:
inbound.jpg

And SIP trunk is OK:
PBX.jpg

I can hear them, they can't. I believe it's an outbound traffic issue.

Also, as I mentioned before, the second NIC is plugged directly to the VOIP provider's router.
The VOIP provider (OTE) claims that their router is completely transparent so no port blocking there. The only firewall that remains is the Debian's firewall.

I don't know if there is something specific to be aware of for 2 NICs PBX configuration.
I mean is the outbound audio routed thru the correct interface?

What do you think I should be looking for?
 
The 3CX firewall test passes successfully.

No gateway is defined for the VOIP interface, as the guide suggests.
 
Hello @Triantafillos Paradas

I would recommend running a wireshark capture of a call and see where the audio is routed. I would assume that the audio is going out the wrong interface and what is the reason you have one way audio. As per the guide the second NIC should not have a gateway defined and static routes must be in place to ensure traffic is routed to the correct NIC
 
Hi @YiannisH_3CX,

I will make a Wireshark capture and I will inform you about. I have no gateway defined for the VOIP interface. Also, I have a static route for the SIP trunk server IP address and I can see (with traceroute) that is routed thru the correct interface.

Is there something else/more that I should add with static routes?
 
If the static route is correct then it should work. Wireshark should be able to tell you more in regards to where the audio is routing.
 
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