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All Inbound Calls Droping When Answered

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cydney

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Good morning. I searched around and found a few issues like this but none seemed to work.

All of the inbound calls to our office fail on our end when answered. The call just drop like we didn't have a call at all. However on the callers end the phone is still connected but with dead silence.

Tried restarting services, restarted the SBC and phones but still nothing.

Outbound calls are not affected though.

Our office is on a SBC since the PBX is hosted in AWS. It has been running fine for months and there was nothing changed in our network or router so I'm lost.

Any and all help is appreciated.

3CX v. Pro 15.5.13103.5

Phones are all Yealink...
- T42S v66.83.0.20
- T46S v66.83.0.20
 
You have upgraded to SP5, did the issue exists in the previous 3CX version

Have you reprovisoned the phones, as new template for these phones - Under Phones, do the extensions show up in red/
 
You have upgraded to SP5, did the issue exists in the previous 3CX version

Have you reprovisoned the phones, as new template for these phones - Under Phones, do the extensions show up in red/

It was happening as soon as I came into the office this morning. 8AM CST. I noticed that the SP5 update was available and applied that. After the update had finished I then went through and updated all the phones to the above version.

The drop out is not limited to the Desktop phones but even Android 3CX app, and softphones on the computers drop the call as soon as it answers.

The caller however seems to stay connected but there is no sound and I do not see the call in the system as an active call any more.
 
I began to check the SBC and the firewall, thought again nothing has changed in the network and everything seems to be fine.

Even connected to 4G on my phone any call I answer drops out.

Firewall check passed as well. I had top open additional port originally I had 9000-9500 open but I notice in the firewall check the number is now 9000-10733.
 
Do you get any error messages for one of these calls in the Activity Log? It sounds as if a SIP message is not getting through.
 
Here are a few of the log out puts.

06/18/2018 2:25:56 PM - [CM503003]: Call(C:33): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5488
06/18/2018 2:25:56 PM - [CM503003]: Call(C:33): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5080
06/18/2018 2:24:14 PM - [CM503003]: Call(C:31): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5080
06/18/2018 2:24:14 PM - [CM503003]: Call(C:30): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5080
06/18/2018 2:24:14 PM - [CM503003]: Call(C:30): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Cancelled/INVITE from 127.0.0.1:5080
06/18/2018 2:24:14 PM - [CM503003]: Call(C:31): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Cancelled/INVITE from 127.0.0.1:5080
06/18/2018 2:24:14 PM - [CM503003]: Call(C:31): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5488
*sanitized SHFT is not my domain.


I do have some good news, when a person calls into our office the call is directed to a call que (Home Office) The reason we did this was if there is no receptionist then one of the other few people in the office can answer the phone. Were not that large of a company in our office.

We also have a small call center that also also has its own Queue and it was suffering from the same issue.

The way the system was setup was that the calls are directed to the Que from there specific Trunk. There is a Trunk for Home Office and a Trunk for the Call Center. If the calls from ether of the Trunks are sent to a Queue then as soon as I pick up the phone the call drops on my end. The Caller is still connected on his phone but no sound. The Caller also drops off of the active calls list when I pick up to answer.

That being said if I set the Trunks to send the call to a Group (Home Office) I can answer and everything is okay. Calls can be answered with no issue.
 
So I spun a new server in AWS, and applied a backup from yesterday and still getting the issue. If I try and send calls from a Trunk directly to a Queue as soon as someone picks up the call it drops on our end.

Just thought I would add that even on on a new install this seems to be happening so now I'm really befuddled.

Right now I just have them ringing to a Group for our home office since there are only four of us.
 
So you've narrowed the issue down to trunk calls to queue members. You have determined that calls from the same trunks work OK when set to other destinations. Which means that it would appear to be a problem with the queue( how calls route, local/remote?), or the sets in the queue.

If you haven't done so, send (route) a trunk call direct to a queue members set, does that work?
Call the queue from a 3CX set that is not a member, does the call drop?
Try answering a trunk call on a non-queue set, then transferring to the queue, or a set that is a member of the queue.

Which of these scenarios works, and which fails. It might help determine where the problem is.
 
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I would also very much like to hear what happens with the tests leejor suggested.

Just to make sure I understand what happens though in the scenario where you have a SIP Trunk sending calls directly to a Queue:
  1. External caller calls into your 3CX and gets forwarded to a Queue
    > Do they hear the Queue Music/WAV file?
  2. The caller remains in the Queue and the call is not answered immediately, lets say they wait in the Queue for 30-40 seconds.
    > Do they continue to hear the Queue Music?
  3. A Queue Agent answers the call eventually.
    > Does the caller and the Queue Agent that answered manage to talk at all, even for a few seconds? Or is the issue immediately apparent (no audio)?
  4. The Agents phone, Yealink or Mobile, when they hear no audio, does it immediately hang up? Or does it seem like there is an active call for some time (lets says ~30 seconds), before dropping?

2 more questions regarding the SIP Trunk:
  • Is it a 3CX supported SIP Trunk provider?
  • In the SIP Trunk settings in the "Options" tab, are the following options like this?
    • PBX Delivers Audio: checked
    • Supports Re-Invite: unchecked
    • Support Replaces: unchecked
 
So you've narrowed the issue down to trunk calls to queue members. You have determined that calls from the same trunks work OK when set to other destinations. Which means that it would appear to be a problem with the queue( how calls route, local/remote?), or the sets in the queue.

If you haven't done so, send (route) a trunk call direct to a queue members set, does that work?
Call the queue from a 3CX set that is not a member, does the call drop?
Try answering a trunk call on a non-queue set, then transferring to the queue, or a set that is a member of the queue.

Which of these scenarios works, and which fails. It might help determine where the problem is.

Q> Call the queue from a 3CX set that is not a member, does the call drop?
A> No the call seems to work fine when called from in office and not an external line.

Q> Try answering a trunk call on a non-queue set, then transferring to the queue, or a set that is a member of the queue.
A1> Set the Trunk to call a Ring Group, Answered calls normally and transferred call to a Users Desktop Phone with no issue.
A2 > Set the Trunk to call Que-1 using an internal phone to make the call. Answered and call dropped. However calling Que-1's extension directly from an internal phone answered fine and transferred fine.

Q> If you haven't done so, send (route) a trunk call direct to a queue members set, does that work?
A1> Set the Trunk to send to Que-1 (Forgot no one was logged in and if no one is there it sends to another Que.) Agent in Que-2 answered with no issues. Called from internal phone.
A2> Set the Trunk to send to Que-1 no issues. Called from internal phone.
A3> Set the Trunk to call me directly and no issues. Transferred to another outside of my Ring Group okay.
 
Last edited:
I would also very much like to hear what happens with the tests leejor suggested.

Just to make sure I understand what happens though in the scenario where you have a SIP Trunk sending calls directly to a Queue:
  1. External caller calls into your 3CX and gets forwarded to a Queue
    > Do they hear the Queue Music/WAV file?
  2. The caller remains in the Queue and the call is not answered immediately, lets say they wait in the Queue for 30-40 seconds.
    > Do they continue to hear the Queue Music?
  3. A Queue Agent answers the call eventually.
    > Does the caller and the Queue Agent that answered manage to talk at all, even for a few seconds? Or is the issue immediately apparent (no audio)?
  4. The Agents phone, Yealink or Mobile, when they hear no audio, does it immediately hang up? Or does it seem like there is an active call for some time (lets says ~30 seconds), before dropping?

2 more questions regarding the SIP Trunk:
  • Is it a 3CX supported SIP Trunk provider?
  • In the SIP Trunk settings in the "Options" tab, are the following options like this?
    • PBX Delivers Audio: checked
    • Supports Re-Invite: unchecked
    • Support Replaces: unchecked

Q> External caller calls into your 3CX and gets forwarded to a Queue. > Do they hear the Queue Music/WAV file?
A> Yes they will hear the announcement (Thanks for calling..) The queue is set to play the whole message before ringing anyone. Music plays but the moment the call is answered the call drops. The Callers phone will stay connected but 3CX will not show any active calls at all.

Q> The caller remains in the Queue and the call is not answered immediately, lets say they wait in the Queue for 30-40 seconds. > Do they continue to hear the Queue Music?
A> Yes music continues to play.

Q> A Queue Agent answers the call eventually. > Does the caller and the Queue Agent that answered manage to talk at all, even for a few seconds? Or is the issue immediately apparent (no audio)?
A> The drop is imitate, there is no time to speak at all.

Q>
The Agents phone, Yealink or Mobile, when they hear no audio, does it immediately hang up? Or does it seem like there is an active call for some time (lets says ~30 seconds), before dropping?
A1> Agents are using Yealink T42S, running FW: 66.83.0.20. The drop out happens as soon as the receiver is lifted from the cradle. The phone screen shows the call and barely have time to tick a 1 sec on the time counter.
A2> Other office Staff are using Yealink T46S, running FW: 66.83.0.20.
A3> They were all updated after updating the PBX to 15.5.13103.5 (v15.5.0)

Q> Is it a 3CX supported SIP Trunk provider?
A> Yes, we use Telnyx for our provider.

Q>In the SIP Trunk settings in the "Options" tab, are the following options like this?
PBX Delivers Audio: checked?
--YES it is checked on all
Supports Re-Invite:
unchecked?
--YES, on the Main Office inbound trunk is was unchecked. --No, on the call center inbound and outbound trunk it was checked.
Support Replaces: unchecked?
--YES this option was uncheck on all.
 
In an effort to, well try anything I could think of. I created a new Inbound and Outbound trunk just like the others are setup.I was using 4x new number allocating 2x numbers for inbound and 2x for outbound.

I then setup the same Inbound and outbound rules just as I already have, other than they are going to a Queue that is not in use by Agents or Office workers. I am the only agent in this Queue.

Test a call from an outside line, call goes into the Queue, rings my desk phone, and I am able to answer the call with no issues. So it looks like its working fine now on the new trunk that I created. Event log showed no errors or issues. Below are the Activity Log for the time of the call, it looks like a whole lot of failing but I do not look at the event log very much.

In short adding the a new trunk for inbound and outbound seems to have fixed the issue. I play on deleting one of the current trunks after office hours and re-adding it and see if this fixes the issue. On the new Trunk I used the settings that NickD_3CX suggested in OPTIONS.

Unless anyone has any other suggestions I will update after I try deleting and re-adding trunks. I'm still wondering what changed from Sunday to Monday. We do have update turned on for Monday morning but the 15.5.13103.5 (v15.5.0) update was not installed and I manually did that. Then updated the phones FW. The issue was present before I ran the 15.5.13103.5 (v15.5.0) update.

>>>>BEGIN LINE>>>>
06/19/2018 12:27:23 PM - [CM503003]: Call(C:100): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5488
06/19/2018 12:27:06 PM - Call to T:Extn:1004@[Dev:sip:[email protected]:5060,Dev:sip:[email protected]:5488;rinstance=93812f00d58b1575] from L:100.1[Queue:8050] failed, cause: Cause: 486 Busy Here/INVITE from 127.0.0.1:5080
06/19/2018 12:27:06 PM - [CM503003]: Call(C:100): Call to <sip:[email protected]:5060> has failed; Cause: 486 Busy Here/INVITE from 127.0.0.1:5080
06/19/2018 12:27:02 PM - Call to T:Extn:1004@[Dev:sip:[email protected]:5060,Dev:sip:[email protected]:5060;rinstance=1-4df9d8e0a7a745459c1a0bf46e680433,Dev:sip:[email protected]:5488;rinstance=93812f00d58b1575] from L:100.1[Queue:8050] failed, cause: Cause: 486 Busy Here/INVITE from 127.0.0.1:5080
06/19/2018 12:27:02 PM - [CM503003]: Call(C:100): Call to <sip:[email protected]:5060> has failed; Cause: 486 Busy Here/INVITE from 127.0.0.1:5080
06/19/2018 12:13:48 PM - [CM503003]: Call(C:97): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5080
06/19/2018 12:13:48 PM - [CM503003]: Call(C:97): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5488
06/19/2018 12:13:48 PM - [CM503003]: Call(C:97): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Cancelled/INVITE from 127.0.0.1:5080
06/19/2018 12:13:48 PM - [CM503003]: Call(C:97): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Cancelled/INVITE from 127.0.0.1:5080
06/19/2018 12:13:48 PM - [CM503003]: Call(C:97): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Cancelled/INVITE from 127.0.0.1:5080
<<<<END OF LINE<<<<
 
After removing the Trunks and setting them up again everything seems to be back to normal. Call come in as they should are sent to there respective Queue and we are able to answer them and they are not dropping out. I'll still be watching this pretty hard for the next few days but all seems well for now. I'll update if there are any more issue with this. Tomorrow will be the real test when its working all day. Thanks

Event log looks fine. Activity log I'm still not sure about.

06/19/2018 5:24:03 PM - Exception: SipMessage::Exception Missing header Contact @ SipMessage.cxx:1393
06/19/2018 5:23:56 PM - Exception: SipMessage::Exception Missing header Contact @ SipMessage.cxx:1393
06/19/2018 5:23:55 PM - Exception: SipMessage::Exception Missing header Contact @ SipMessage.cxx:1393
 
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