No Outgoing Audio for 1 of the SIP Trunk (Connect to ISP IMS Server via VLAN)

Router used: RB4011iGS

MY ISP, Unifi- TM in Malaysia comes with triple play service, Internet, IPTV and Voice, where each service has its own VLAN.

My Rasberry Pi hosting FreePBX has 2 network interfaces. One of them is connected to the normal VLAN (500) where it is accessible from the local network whereas the other one is connected to a separate interface in the Mikrotik that is tagged under the Voice VLAN

I am using chansip to connect to my ISP VOIP Server. Below are my configurations

[email protected]

Register String: [email protected]:123456:[email protected]@

The problem here is when I go to Astriek Info. This trunk has Status OK and the SIP Account is registered successfully.

I have made a few test calls. I am able to receive incoming and made outgoing calls. However, I only able to receive incoming audio, but the other party is not hearing what I am saying.

I have tried other SIP Trunks, but did not face such problems

If you have two connections contending for the same port, the oldest registration will fail it’s next call of course the media will also go to the wrong place also

So, can you enlightened me on how I should resolve the problem? By the way, can I confirm if you understand what I am saying?

Which part of the configuration is saying two connections are contending for the same port ?

Although I don’t think it really explains your symptoms, please replace this by:


Also consider why you need G.729 and both variants of G.711, assuming that Malaysia aligns with Europe, rather than the USA and Japan, I’d suggest:


If you have no outbound audio, it is possible that you need nat=comedia (an explicit nat=no is almost never needed).

Generally it is helpful to have the full log resulting from the use of the CLI command “sip set debug on”, when diagnosing one way audio, as the SDP contents can be useful.

(type=friend is an unnecessary security risk. username is only relevant with host=dynamic It is better to have an explicit dtmfmode. The official name for canreinvite has been directmedia for a very long time. New installations should use chan_pjsip.)


I have found out the problem. I resolve the issue my routing the entire subnet in my Raspberry Pi to the second interface that is connected to VLAN400.

Voila, now it able to receive and make multiple simultaneous calls which is not possible to do so if the phone is connected directly to my ONU

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