Loss of Outbound Calls

[Richard Marsh]

Our SIP trunk provider moved us to a new platform a couple of weeks ago. Since, we have been experiencing line drops every 15 minutes on inbound calls.

We spoke with the support team yesterday as I suspect that the new platform is sending a 'INVITE' request every 15 minutes and doesn't get a 'ACK' response and therefore sends 'BYE' response terminating the call.

We received an update from our trunk providers support claiming that they have corrected the issue. Unfortunately, we can not check this as we have lost all outbound call ability. Our Event log is showing the following error:

Call or Registration to +44XXXXXXXXXXX@(Ln.10001@EntaNet SIP Trunk) has failed. XXX.XX.XX.80 replied: 500 Service Unavailable; from IP:XXX.XX.XXX.XX:5060

Our service provider claim that to be an internal PBX error, which error 500 is but, is it returning this error because some mismatch configuration on the supplies end?

 

[eddv123]

Hi Richard Marsh,

A 5xx response is a server error response. For some reason the server cannot fulfil the request. Coming from the IP (that you have blanked out) - what is this IP address ? is it the provider by any chance ?

Who is the provider by the way ? are they supported ? https://www.3cx.com/partners/sip-trunks/

 

[Richard Marsh]

Its the providers subnet address.

To update - after escalating this to a senior manager, following total outage of the trunk, we now have both inbound and outbound calls working again. With that said, we are still experiencing the original error of calls dropping every 15 minutes on in bound traffic.

Entanet, and no they are not on the list of tried and tested by 3CX, but have used them for 8 years with no real issues.

 

[YiannisH_3CX]

Hello @Richard Marsh

The 15 minute mark would indicate a session timer expiring so i would start by looking into that. If you are familiar with wireshark run a capture and check if there is a re-invite at the 15 minute mark, who is sending it and what is the reply.
What endpoints are you using? Are you using IP phones or 3CX clients?

 

[Richard Marsh]

Hi Yiannish

Thank you for your posting.

I have run wireshark and have noted that the provider is sending the 'INVITE' followed by 'BYE' at 15 minutes. We are using mixed, both IP phones and 3cx clients. This is an issue that only recently started when the provider switch us from their old platform to the new replacement one.

Both inbound and outbound calls are working again, with no adjustments our end. I am fairly confidant that this is with the provider not our local PBX as our VoIP Unlimited trunk works fine.

 

[YiannisH_3CX]

We have many of our supported provider that do update the session after a certain time period without the call being dropped and technically is not wrong to do so. The provider could use a re-invite to achieve this or an Update message. If this is implemented correctly on both sides then the call should continue without any issues and it does with our supported providers.
Make sure that your endpoints are updated to the supported for the PBX version and try again. If the issue persists then you should take this up to your provider and present them with the capture of the call. They should be able to fix the issue. if not then i would recommend moving to a supported provider.

 

[Richard Marsh]

Hi YiannishH

We have heard from the provider, Entanet with the following response to the issue.

Following our conversation on the phone we have done some investigating and we can see that the cause of the drops is down to an issue in the signalling chain for the call.

Taking the call id [email protected]X as an example we have the normal invite signalling, but we also send a publish which is not handled correctly by your PBX. We receive a proxy auth required instead of a 200 ok.

The below show's what happens, as our proxy doesn't receive the correct response, therefore, the signalling chain is invalidated and the call drops.


If you have any queries please let us know.

Session Initiation Protocol (PUBLISH)
    Request-Line: PUBLISH sip:[email protected] SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 195.74.100.80:5060;branch=z9hG4bKb80c.564b5b57.0
        To: sip:[email protected]
        From: <sip:[email protected]>;tag=dc708c7d6d3adbddbb044804c052ee7d-8ba3
        CSeq: 10 PUBLISH
        Call-ID: [email protected]
        Max-Forwards: 70
        Content-Length: 566
        User-Agent: Entanet UAC V8.24
        Event: dialog
        Expires: 7201
        Content-Type: application/dialog-info+xml
    Message Body
        <?xml version="1.0"?>\n
         [truncated]<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" state="partial" entity="sip:[email protected]"><dialog id="cf0cd8a5-f885-1236-fba5-fa163e2e6107" call-id="cf0cd8a5-f885-1236-fba5-fa163e2e6107" local-tag="588ff104


Session Initiation Protocol (407)
    Status-Line: SIP/2.0 407 Proxy Authentication Required
    Message Header
        Via: SIP/2.0/UDP 195.74.100.80:5060;branch=z9hG4bKb80c.564b5b57.0
        Proxy-Authenticate: Digest nonce="414d53595b39fed108:674331b4a9b36b0645b41574fffff93d",algorithm=MD5,realm="3CXPhoneSystem"
        To: <sip:[email protected]>;tag=d0a2bc4d
        From: <sip:[email protected]>;tag=dc708c7d6d3adbddbb044804c052ee7d-8ba3
        Call-ID: [email protected]
        CSeq: 10 PUBLISH
        User-Agent: 3CXPhoneSystem 15.5.13103.5 (12071)
        Content-Length: 0

Any thoughts?

 

[YiannisH_3CX]

Please note that when an Invite is sent to the provider the Invite message contains an Allow Header which includes all the actions the other party can use during the call. The party that receives the Invite sends its own Allow header in the 200 OK message.
3CX PBX does not allow the PUBLISH sip message as it is not included in the Allow header so the provider should not send it.
This is what the 3CX PBX currently supports:

You could in this scenario ask the provider why they are sending PUBLISH since it is not included in the Allow header but i would recommend switching to a supported and tested provider.
https://www.3cx.com/partners/sip-trunks/