3CX auto attendant playback problem!

[Don_Zalmrol]

Hello all,

As the title says, our 3CX server will not play any auto attendant message for external callers or even the voicemail options.

Yesterday all of a sudden, the 3CX server gave me a notice that we where running the "free version" of the system.
Which is not possible, since we payed for the licensed and activated it ourselves…

I've inserted the registration key again, and now it's active again. But now the auto attendant/ voicemail doesn't work anymore externally…

Here is the situation:

When I dial on a local sip phone to auto attendant 1 (number 800) I hear my voice with the options.

When I dial the line number with my cell phone or a neighbours line phone, the message won't play.

The strange part is that this worked perfectly for almost 3 weeks now and it just stopped.
The 3CX is a licensed version (3CXPSSB).

When a call is placed it is routed towards the auto attendant, and I only hear a phone dial.
Enclosed is a snapshot of the server log, where you can see that the incoming call is properly forwarded to the 804 which is the out-of-office auto attendant,
instead of the 800 AD which is the normal workday AD.

I've already contacted our supplier/ installer for our 3CX system, but they are also puzzled and also searching for a solution.



Here is my problem in a nutshell:
- Internally calls toward to auto attendants groups/ ring groups or voicemail == No problem
- Externally == nothing, only an active dialing tone (like you are being transferred)…

Can anybody help me?

Many thanks in advance & sincerely,

Laurens

 

[craigreilly]

The ACK is not received might be problamatic. I do not see an ACK in mine - it might be assumed,
Yours is clearly showing no ACK.

If you are able - https://www.callcentric.com/login/
gives you a 777 area code phone number. I have an account here to then I could call your system.
This would rule out an issue with your current VOIP Provider - maybe.

 

[Don_Zalmrol]

Well the thing is, that I have also a simbox (which works like a cellphone).
This is configured and when I call the number of simcard, I hear the auto attendant.

With the link you provided, I can use one free telephone number starting with 777. Correct?

00:45:42.332  [MS105000] C:111.1: No RTP packets were received:remoteAddr=10.0.0.10:20000,extAddr=0.0.0.0:0,localAddr=10.0.0.2:7430
00:45:40.966  [CM503003]: Call(111): Call to sip:[email protected] has failed; Cause: 487 Request Terminated; from IP:10.0.0.16:5062
00:45:40.856  [CM503008]: Call(111): Call is terminated
00:45:33.816  [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Yealink SIP-T38G 38.0.129.4] PBX contact: [sip:[email protected]:5060]
00:45:33.816  [CM503002]: Call(111): Alerting sip:[email protected]:5062
00:45:33.494  [CM503025]: Call(111): Calling Ext:Ext.100@[Dev:sip:[email protected]:5062]
00:45:33.445  [CM503004]: Call(111): Route 1: Ext:Ext.100@[Dev:sip:[email protected]:5062]
00:45:33.445  [CM503010]: Making route(s) to <sip:[email protected]:5060>
00:45:20.373  [CM503007]: Call(111): Device joined: sip:[email protected]:40600;rinstance=355f3410eb7b0356
00:45:20.370  [CM505001]: Ext.999: 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]
00:45:20.370  [CM503002]: Call(111): Alerting sip:[email protected]:40600;rinstance=355f3410eb7b0356
00:45:20.218  [CM503025]: Call(111): Calling Ext:Ext.999@[Dev:sip:[email protected]:40600;rinstance=355f3410eb7b0356]
00:45:20.165  [CM503004]: Call(111): Route 1: Ext:Ext.999@[Dev:sip:[email protected]:40600;rinstance=355f3410eb7b0356]
00:45:20.165  [CM503010]: Making route(s) to <sip:[email protected]:5060;user=vmail>
00:45:17.562  Currently active calls - 1: [111]
00:45:17.466  [MS211000] C:111.1: 10.0.0.10:20000 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.
00:44:58.225  [CM503007]: Call(111): Device joined: sip:[email protected]:40600;rinstance=e22c1ae96fdd8c87
00:44:58.225  [CM503007]: Call(111): Device joined: sip:[email protected]:5060
00:44:58.221  [CM505001]: Ext.804: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:[email protected]:5060]
00:44:58.221  [CM503002]: Call(111): Alerting sip:[email protected]:40600;rinstance=e22c1ae96fdd8c87
00:44:58.071  [CM503025]: Call(111): Calling Ext:Ext.804@[Dev:sip:[email protected]:40600;rinstance=e22c1ae96fdd8c87]
00:44:58.027  [CM503004]: Call(111): Route 1: Ext:Ext.804@[Dev:sip:[email protected]:40600;rinstance=e22c1ae96fdd8c87]
00:44:58.027  [CM503010]: Making route(s) to <sip:[email protected]:5060>
00:44:58.026  [CM505002]: Gateway:[Simbox] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Mv-37x (1008030)] PBX contact: [sip:[email protected]:5060]
00:44:58.025  [CM503001]: Call(111): Incoming call from 10002@(Ln.10002@Simbox) to <sip:[email protected]:5060>
00:44:58.020  [CM503012]: Inbound out-of-office hours rule (unnamed) for 10002 forwards to DN:804

 

[craigreilly]

that is right. i paid nothing for the service.
i had free calls to any 777 number.
just not sure if it works outside of North America.

 

[Don_Zalmrol]

I will give it a try, but I live in Belgium.
So here goes nothing.

I will route you to the same auto attendant, since everyone in my house is sleeping at the moment.
And if the phone rings, there will be hell to pay

 

[craigreilly]

in case you need it:
http://www.callcentric.com/support/device/3cx

 

[Don_Zalmrol]

The new provider won't registrate…
I did everything as the guide instructed.

http://www.callcentric.com/support/device/3cx

I apparently I get a DNS error

Registration at CallCenric has failed. Destination (sip:[email protected]:5060) is not reachable, DNS error resolving FQDN, or service is not available.

 

[craigreilly]

try changing fqdn to ip: 204.zzzzzzz

 

[Don_Zalmrol]

By that you mean the outbound proxy hostname?

EDIT: Still no progress. I think it's not possible from end, or have the wrong FQDN for my end in the world.

 

[craigreilly]

proxy and sip server host both.

 

[Don_Zalmrol]

Changed both of them to the FQDN you've provided me.
Alas, still the same problem

Registration at Test has failed. Destination (sip:xxxxxx@yyyyyyyy:5060) is not reachable, DNS error resolving FQDN, or service is not available.

 

[craigreilly]

must be them blocking it...
Sorry - my VOIP line is for Continental US Calls only and I have no other lines provisioned at the moment...

 

[Don_Zalmrol]

No, problem.
You already helped very well.

Tomorrom I'm going to call my VOIP Provider.
The problem is with them or in my router…

Thank you for your help & goodnight!

Laurens

 

[Don_Zalmrol]

Today the 3CX installer from my VOIP retailer came by and he found the problem!

Apparently the problem comes from my new network card that uses two network ports as one teamed up card to improve bandwidth and throughput.

There goes something wrong when 3CX (or Windows) tries to stream audio through that virtual card (the team)…

Now I'm looking into it to re-enable the team without disabling 3CX!

I will keep this post updated for others who might have the same problem.

And thank you for your help Craig Reilly!

Cheers,

Laurens

 

[craigreilly]

Who wouldda thought...
Ive only run into a teaming problem once . I Think it was on a Citrix server.

Good luck.