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IP phones have got no audio but only on outgoing calls

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BradStarrate

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Hi,

Please could someone maybe assist with this weird issue.

The handsets that connect to a dect when they call out have no audio. We can not hear them and they can not hear us but it is only the handsets which are W56H handsets.
The W56H's connect to W80B base stations multi-cell which we have restarted. Internal calls for these handsets are working fine as well
The 3CX is hosted at Azure and that we are running a Mikrotik RB2011. The firmware that we running on the dects are the following 103.83.0.75 (Firmware version)
We also have yealink T46S’s are working fine and internal calls are working fine as well.
But if you transfer a call from the T46S to a cordless phone after first making the outbound call from the T46S then the W56H works fine. The client's issue only happens when you initiate the cal from the W56H, then there is 0 audio as stated above.
We have asked the ISP if there are any issues where there are not. Plus we checked on our Mikrotik if there is anything being blocked and all ports are allowed.
 
My guesses would be either a Codec issue, or a port issue. Are these devices behind an SBC, or using STUN?
Have you checked the 3CX Activity Log, for any information, about one of the calls, that may help?
 
We have STUN and changed the codecs from G729 to G.711 U-law reprovisioned the phones after the codecs were changed and still problem persists.
 
Also the ports for RTP and SIP are the following

RTP Ports: min - 14140 / Max - 14740

SIP ports: 5068

Hope this helps
 
Is the option PBX Delivers Audio enabled, for the extension(s)?
 
indeed it is enabled.
 

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One from me, on the mikrotik have you turned off the sip service port?
 
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I am having the same issue... Did you resolve?
 
These are sometimes hard things to debug. I can see a few potential issues that others already have addressed. We use the MT2011 a lot, so i know for a fact that this is not the limiting factor. Please DISable SIP ALG in the MT: https://www.maxo.com.au/support/troubleshooting-articles/sip-alg/disabling-sip-alg--mikrotik-routers
SIP ALG may mess up the RTP ports where 3CX is sending the audio to (and it is, because PBX Delivers Audio is enabled).

Furthermore: the RTP ports your trunking provider uses may not be the RTP ports your PBX uses. So please check whether outbound you can use the UDP portrange your trunking provider requests. Personally i usually add a f/w rule outbound UDP to the ip range of the trunking provider. That way if something changes on the far end, i don't have to worry about it.

Also: please mind that 711.u is used outside of Europe. If you live in Europe you better use 711.a. The difference basically is how the codecs encodes audio with very low and very high volumes. If your (EU/US) trunking provider is going to recode back and forth you are going to loose (audible) audio quality in these area's (we have all heard that at one point!)
 
I am having the same issue... Did you resolve?
If you are also using STUN on this model, the please refer to my previous answer so you don't have to wait for confirmation.

When done improperly, it's typical to get one way audio in STUN
 
Dear fellow techs,

I've spent the better part of a few days resolving this. We initially has issue with the mikrotik router firewalling the 3CX server, then after getting thru that, we had issues with one way audio - only to the yealink ip phones, not with the web or smartphone client. Here's what we did to resolve.

Initially I mapped all TCP and UDP ports from the public FQDN ip to the private ip. 5060, etc per the 3cx firewall ports. Where mikrotik failed was above 9999 ports. which then in turn disabled the 3cx media server.

the server fix: I went to a 1:1 NAT config. I'm using a loopback address to assign a single IP (hairpin NAT) on my firewall. Previously I mapped each port from public address to private address. (We currently use ospf to distribute that connected route throughout the network even though it's a /32. I'm not including any of that config but the below commands are just forwarding traffic in and out thru the public loopback IP which is then routed to the rest of the world)

Here are the exact firewall commands on 3cX server connected mtik router I've replaced private ip with 10.0.0.84, and public with 123.456.789.001 (our 3cx instance is running on hyper-v server and picks up DHCP from a port on the router thru ms virtual switch connected to the rb110 (ether11-sw12 will need to be changed to your upstream facing port for that router. might be able to remove out-interface, but I've not tested that.)

add action=src-nat chain=srcnat comment="3CX NEW" out-interface=ether11-sw12 src-address=10.0.0.84 to-addresses=123.456.789.001
add action=dst-nat chain=dstnat comment="3CX NEW" dst-address=123.456.789.001 to-addresses=10.0.0.84

note: also fully disabled sip helper completely on this router/firewall.

So then we were able to get 3cx server to pass the firewall test which previously failed the full cone test.

Now had one way audio from the sip, but calls inbound answered with the app or web worked fine.

on to the Mikrotik router on the test bench:

WAN IP - dynamic from Access point. (can ping fqdn.3cx.us (the loopback public on the RB1100))
Masq nat - 192.168.1.x lan
ip phone DHCP 192.168.1.193

Connects.. no audio RX

the solution:

disabling the sip helper didn't work. same issue.
Disabling the sip direct media worked but leaving the sip helper service enabled.

/ip firewall nat
add action=masquerade chain=srcnat out-interface=wlan1
/ip firewall service-port
set sip ports=5060,5061,5065,5066 sip-direct-media=no


note: We've also set connections to go thru the server rather than direct connect on 3cx server and we use the fqdn for provisioning.

Now onto other projects (finally)

cheers, Andy
 
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Yes.. was the same issue he was having. But the solution was pretty specific on the Mikrotik router, and so I posted a detailed description of what solved the issue for us given the circumstances of our installation.
 
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