VOIP phones inbound calls not working

[Vali_Bogdan]

Hello everyone,

We are having some issues with VOIP  Cisco Spa 303 phones inbound calls. Outbound calls work without any issues. The appliance that we are using is a MX100.  The phones are hosted by a VOIP service provider and we don't have PBX server.

We have been in touch with our VOIP service provider and they said we might have SIP ALG enabled and that's why we have issues. Then we talked with your ISP and they said SIP ALG is disabled.

I know that Meraki MX doesn't use SIP ALG and if you want to use VOIP with Meraki you will need to check the traffic shaping rule. We went trough the documentation https://documentation.meraki.com/zGeneral_Administration/Tools_and_Troubleshooting/VoIP_on_Cisco_Mer...

The phones worked perfectly for a few years before this issue and nothing has changed on the traffic shapping rule side or on our internal network. Making a packet capture of a phone that doesn't work when somebody calls inbound points that the Meraki MX 100 sends the message 404 not found.

We already have a VLAN VOIP for the phones and added the traffic shaping rules.Port forwarding is not a solution as we have multiple phones. I would like some other suggestions or ways on how to fix the issue as we are stuck right now and it seems the issue is within our network.

Many thanks,

Cheers,

Vali

[PhilipDAth]

The MX does not send 404's - something else is doing that.  I'm guessing it was broken before, and is still broken, and is not related to your core issue.

Traffic shapping rules will have no impact on all inbound calls breaking.  This is an unrelated issue.

If only inbound calls are not working then it will be something to do with the processing of NAT traffic.

Do the phones ring, you answer, and they fail to get voice working?  Or do the phones not ring at all?

For all the phones we do we use SIP/TLS.  Being TLS the entire call setup is encrypted.  Consequently ALG's can not see the traffic or interfere with it.  Also it allows the CPE to track the session because it sits on top of TCP, rather than UDP.

If you can't use SIP/TLS then the second option I recommend you try is SIP/TCP.

[Vali_Bogdan]

Hello Philip,

Thanks for clarifying my statements from above. Many thanks for the quick reply. The phones don't ring at all it just says the standard message "not available to take your call please leave a message after the tone" and then it goes directly to voicemail. The phones are registered with the VOIP service provider.

Currently we don't have any rules on NAT.

Cheers

[Nick]

Do you know what method the phones are using to stay registered with the PBX?

Is there a PBX interface where you can see which extensions are online etc.

You could attempt to forward SIP traffic and audio directly to a phone to confirm if it is the gateway that is causing the issue.

Do you have anything upstream from the MX100?

[Vali_Bogdan]

Hello Nick,

The PBX server is hosted by our VOIP service provider and we as users have a Administrator interface where we can add terminals to numbers. I have attached a printscreen with the view.

Upstream from MX100 is the Cisco 2900 router from the ISP.

Cheers