ACK is not received Error.

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Hi Thanks for the feedback.

If you have a support ticket already open, then better continue dealing with support. More assistance can be provided through proper channels.

Thanks
 
Will do. My post here was really just to let others who might still be struggling with the issue that I had found the bug and it was being worked on.

Thanks for your help. Dom.
 
For all people experiencing this problem here is a solution.

If there is anything listed in the Settings > Nextwork > STUN Server > "Public IP to specify in Contact and SDP" field then the 3CX system will use this in the SIP "Contact" message regardless of whether you have STUN turned on or off.

The trick is to remove the value from this field and this is done as follows.

Go into Settings > Advanced > Custom Parameters

Highlight the row for "MSEXTERNALADDRESS" and press the "Delete" key on your keyboard then repeat and delete the row for "CMEXTERNALADDRESS". you will notice the "Value" field for both these rows is the same as the greyed out field in the other window.

Restart the 3CX services and check these rows no longer appear in your custom parameters. Also check the greyed out field in the STUN window is blank.

3CX are looking to fix this bug long term but this is an effective work around. Any problems drop me a note on this forum.

Regards, Dominic.
 
Hi Dominic,

I'm having the same issue, I was running an older version 8 of 3CX and was having call license errors, so I was advised to upgrade to the latest version.

I have now done this, and now I'm getting ACK errors after about 30 seconds into a call when it cuts out.

Other than upgrading to the latest software I don't see how this can be an issue with our hardware, despite 3CX tech support telling me it's a firewall issue.

It's driving me insane, not to mention my customer!
 
what firewall are you using and what is your configuration setup?

Are you using STUN or not?
 
Hi

we've got a Zyxel Prestige 792h router, 3cx has STUN turned off

calls are absolutely fine, good quality, until ACK error occurs and drops the call, surely that's a major bug?
 
are you running a public IP address on your 3CX server or is your firewall doing NAT?
 
Public IP, but firewall is doing NAT
 
Hi, sorry to sound dumb but I'm not sure I follow.

You would normally have a private IP (192. or 10. or 172.) on the 3CX server with a public IP on the Firewall with NAT

OR....

you would have a public IP on the 3CX server with the firewall not doing NAT

You seem to have a bit of both?
 
Sorry no, the router has the publc IP and the server has a local IP
 
I have the same scenario, and I always find it is best to use STUN in this circumstance as it just seems to sit better with NAT in the router.

Why don't you try enabling the STUN servers and setting them to stun.3cx.com and stun2.3cx.com, ensuring you have the "Keep Alive" option set in one of the parameter windows to ensure you server does not drop the session if it hits it's NAT timeout and see if that is any better for you?

Who is your VOIP provider and is your router just routing up one line?

Cheers, Dom.
 
Has anyone fixed this problem yet. I am getting ACK not received error and calls terminating after 30 seconds only on one provider (voip-unlimited), I dont get any issues with sipgate.

I have updated to the latest version of 3CX, disabled firewall to rule it out and still getting the same problem. I have tried it with STUN off and on using the 3cx.stun.com address, sipgate stun address and the voip-unlimited stun address and still getting the same problem.

Anyone fixed this problem, please post your config and any changes you made.

Help!

Thanks
 
Out of interest, in the field Settings> STUN Server > Public IP to specify in Contact and SDP - do you have anything set. I know the field is greyed out if "Turn off STUN server" is not ticked but regardless of if it is greyed out or not is there anything set in there? Cheers, Dom.

PS. If so, what?
 
Hi,
For Voip-Unlimited to work we have STUN turned off, and the public IP address entered in network settings - STUN server. Have you tried that?

Kind regards
Richard Crabb
 
No there is nothing in this field and it is greyed out. I have been in the custom parameters as a previous forum post stated to look for MSEXTERNALADDRESS and CMEXTERNALADDRESS settings which I have deleted.

This is what gets logged when I call a voip-unlimited number. As stated before the sipgate numbers on my system are fine it is just voip-unlimited.

09:38:53.296 [CM503008]: Call(10): Call is terminated
09:38:53.296 [CM503021]: Call(10): ACK is not received
09:38:25.218 Currently active calls - 1: [10]
09:38:24.781 [MS105000] C:7.1: No RTP packets were received:remoteAddr=91.151.2.133:51026,extAddr=213.131.120.86:53314,localAddr=myexternalip:53314
09:38:23.359 [MS105000] C:8.1: No RTP packets were received:remoteAddr=91.151.8.181:50690,extAddr=213.131.120.86:53316,localAddr=myexternalip:53316

Thanks
 
ok, you're in luck all my lines go via VOIP Unlimited and I don;t have any issues at all so here is my configuration for the VOIP provider as that is probably the best place to start. Clearly your network setup is working otherwise you would be seeing similar issues with sipgate.

VOIP Provider > VOIP Unlimited

General Fields > Provider Details > sip.voip-unlimited.net, 5060, blank, 5060
Advanced > Provider > Re-invite="N", Replace="N", PBX Delivers="Y"
Registration > 600, In and Out, External (STUN Resolved)
Codecs > G729, G.711(U), G.711(A)
Outbound > To.User="Y", Request Line="Y", RemotePartyID="N", No custom settings at all
Inbound > nothing in here
Source ID > Match any fields and not much else, Source Identification by DID = "Y", SIP field containing DID number = Request Line URI: User Part

I have masks as 0123456* for example to identify my DID but you can enter them manually in here.
DID > List of all my DID numbers.

See if you have anything different.
 
This was the only part I had nothing set, however it hasn't made any difference.

Source ID > Match any fields and not much else, Source Identification by DID = "Y", SIP field containing DID number = Request Line URI: User Part

I am a bit confused with the firewall setup as voip-unlimited are insisting I check this again. I have run the firewall checker and get this response

<15:02:47>: Phase 1, checking servers connection, please wait...
<15:02:47>: Stun Checker service is reachable. Phase 1 check passed.
<15:02:48>: Phase 2a, Check Port Forwarding to UDP SIP port, please wait...
<15:02:48>: UDP SIP Port is set to 5060. Response received WITH TRANSLATION 10482::5060. Phase 2a check passed with WARNINGS. Some functionality will be LIMITED. For more information, please visit http://www.3cx.com/blog/docs/firewall-checker/

<15:02:48>: Phase 2b. Check Port Forwarding to TCP SIP port, please wait...
<15:02:49>: TCP SIP Port is set to 5060. Response received WITH TRANSLATION 10482::5060. Phase 2b check passed with WARNINGS. Some functionality will be LIMITED. For more information, please visit http://www.3cx.com/blog/docs/firewall-checker/

<15:02:49>: Phase 3. Check Port Forwarding to TCP Tunnel port, please wait...
<15:02:49>: TCP TUNNEL Port is set to 5090. Response received WITH TRANSLATION 10485::5090. Phase 3 check passed with WARNINGS. Some functionality will be LIMITED. For more information, please visit http://www.3cx.com/blog/docs/firewall-checker/

<15:02:49>: Phase 4. Check Port Forwarding to RTP external port range, please wait...
<15:02:56>: UDP RTP Port 9000. Response received WITH TRANSLATION 10486::9000. Phase 4-01 check passed with WARNINGS. Some functionality may be IMPAIRED. For more information, please visit http://www.3cx.com/blog/docs/firewall-checker/
<15:02:56>: UDP RTP Port 9001. Response received WITH TRANSLATION 10487::9001. Phase 4-02 check passed with WARNINGS. Some functionality may be IMPAIRED

and so on...


Should I be just forwarding 5060 and 9000-9049 to the internal IP address of my 3CX phone system?
 
what firewall are you using?
are you using the same king of SIP setup with SIPGate?
Is this a problem with incoming, outgoing or both lines?
can you turn on a full verbose debug and paste the file with just a single call?
 
I am using a dedicated ADSL broadband line for the 3cx phone system with a managed checkpoint firewall.

16:04:42.000 [CM503008]: Call(6): Call is terminated
16:04:41.984 [CM503021]: Call(6): ACK is not received
16:04:41.421 Currently active calls - 1: [6]
16:04:09.937 [CM503007]: Call(6): Device joined: sip:[email protected]:40600;rinstance=78135f69701562d6
16:04:09.921 [CM503007]: Call(6): Device joined: sip:myphonenumber@sip.voip-unlimited.net:5060
16:04:09.890 [CM505001]: Ext.555: Device info: Device Identified: [Man: 3CX Ltd.;Mod: Voice Mail Menu;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Voice Mail Menu] PBX contact: [sip:[email protected]:5060]
16:04:09.890 [CM503002]: Call(6): Alerting sip:[email protected]:40600;rinstance=78135f69701562d6
16:04:09.687 [CM503025]: Call(6): Calling Ext:Ext.555@[Dev:sip:[email protected]:40600;rinstance=78135f69701562d6]
16:04:09.671 [CM503004]: Call(6): Route 1: Ext:Ext.555@[Dev:sip:[email protected]:40600;rinstance=78135f69701562d6]
16:04:09.671 [CM503010]: Making route(s) to <sip:[email protected]:5060;user=vmail>
16:04:09.656 [CM505003]: Provider:[voipunlimited] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:myphonenumber@myexternalip:10580]
16:04:09.640 [CM503001]: Call(6): Incoming call from 0191xxxxxxx@(Ln.10008@myphonenumber) to <sip:[email protected]:5060;user=vmail>
16:04:09.468 [CM503012]: Inbound office hours rule (unnamed) for 10008 forwards to VM:100
 
what is extension 555?
 
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