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Incoming Rings and immediately goes to voicemail (SOMETIMES)

Discussion in '3CX Phone System - General' started by Irving, Jan 18, 2018.

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  1. Irving

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    I'm having a weird issue i can't manage to figure out. About a quarter of the incoming calls are quickly ringing (about half a second) and sends the caller to voicemail. If you call again it may hit the ring group as normal. I'm using the same VOIP provider however i'm using their sip peering function (essentially disabling their control panel) with a new 3cx install.

    Any help would be greatly appreciated.

    Thanks,
     
  2. Irving

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    Just for kicks i put an IVR (set the timeout to 2 seconds) in front of the Ring Group and haven't had a problem with calls going to voicemail.
     
  3. leejor

    leejor Well-Known Member

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    Did you check the Activity Log right after one of the calls to see why it went to voicemail so quickly. The reason should show in the log.
     
  4. YiannisH_3CX

    YiannisH_3CX Support Team
    Staff Member 3CX Support

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    Hello @Irving

    I would check if a member of the ring group has a second device / Client registered somewhere that may answer / forward / reject the call. There is also a chance that an extension might have the option "Ring my mobile simultaneously" enabled and the call is forwarded to that.
     
  5. Irving

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    I'm getting alot of this in the low log setting.

    01/19/2018 10:02:22 AM - [CM503003]: Call(C:925): Call to <sip:110@10.1.1.70:5060> has failed; Cause: 487 Request Cancelled/INVITE from 192.168.0.102:5060
    01/19/2018 10:02:22 AM - [CM503003]: Call(C:926): Call to <sip:101@10.1.1.70:5060> has failed; Cause: 487 Request Cancelled/INVITE from 192.168.0.124:5060
    01/19/2018 10:00:56 AM - [CM503003]: Call(C:920): Call to <sip:110@10.1.1.70:5060> has failed; Cause: 487 Request Cancelled/INVITE from 192.168.0.102:5060
    01/19/2018 10:00:56 AM - [CM503003]: Call(C:920): Call to <sip:102@10.1.1.70:5060> has failed; Cause: 487 Request Cancelled/INVITE from 192.168.0.101:5060
    01/19/2018 10:00:42 AM - [CM503003]: Call(C:921): Call to <sip:101@10.1.1.70:5060> has failed; Cause: 487 Request Cancelled/INVITE from 192.168.0.124:5060
    01/19/2018 10:00:42 AM - [CM503003]: Call(C:919): Call to <sip:110@10.1.1.70:5060> has failed; Cause: 487 Request Cancelled/INVITE from 192.168.0.102:5060
    01/19/2018 10:00:15 AM - [CM503003]: Call(C:917): Call to <sip:101@10.1.1.70:5060> has failed; Cause: 487 Request Cancelled/INVITE from 192.168.0.124:5060
    01/19/2018 10:00:15 AM - [CM503003]: Call(C:917): Call to <sip:102@10.1.1.70:5060> has failed; Cause: 487 Request Cancelled/INVITE from 192.168.0.101:5060
    01/19/2018 9:59:54 AM - [CM503003]: Call(C:915): Call to <sip:110@10.1.1.70:5060> has failed; Cause: 487 Request Cancelled/INVITE from 192.168.0.102:5060
    01/19/2018 9:59:54 AM - [CM503003]: Call(C:916): Call to <sip:101@10.1.1.70:5060> has failed; Cause: 487 Request Cancelled/INVITE from 192.168.0.124:5060
    01/19/2018 9:59:49 AM - [CM503003]: Call(C:914): Call to <sip:110@10.1.1.70:5060> has failed; Cause: 487 Request Cancelled/INVITE from 192.168.0.102:5060
    01/19/2018 9:59:49 AM - [CM503003]: Call(C:913): Call to <sip:101@10.1.1.70:5060> has failed; Cause: 487 Request Cancelled/INVITE from 192.168.0.124:5060
    01/19/2018 9:59:33 AM - [CM503003]: Call(C:910): Call to <sip:101@10.1.1.70:5060> has failed; Cause: 487 Request Cancelled/INVITE from 192.168.0.124:5060
     
  6. leejor

    leejor Well-Known Member

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    How is your network set up? You have IP 10.1.1.70 showing, yet IPs with 192.168.0.XX, different range altogether.

    Who do these IPs belong to? Is the 3CX server local to all sets?
     
  7. JCLloyd

    JCLloyd New Member

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    Those are sent to the extensions in the ring group that did not answer, after one did. By process of elimination, note all the extensions that got the invite cancel, and you should be missing the one that answered. That would be the extension to check for call redirection while logged in. My guess would be that it is close to being the first number listing in the ring group.
     
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  8. Irving

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    192.168.0.0/24 is branch with yeahlink phones.
    10.1.1.0/24 is Amazon where we are VPN into where all ports are opened through the VPN.
     
  9. Irving

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    Ah ok makes sense.
     
  10. Irving

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    This is a new install and I wanted to keep everything simple so i did not add any mobile numbers and left most of the ext options in their defaults.
     
  11. Irving

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    Not sure if this is related but now external branches cannot hear each other when they dial by via an extension (ie: 101 calls 201). It works inside the branch though (ie: 101 calls 105). See branch design below:

    Branch 1: 192.168.0.0/24 with extensions 100-150
    Branch 2: 192.168.5.0/24 with extensions 200-250
    Branch 3: 192.168.128.0/24 with extensions 300-350

    That said and what makes this puzzling is that if they dial their DID's that works just fine (ie: Someone from Branch 1 calls DID in Branch 2 and that works.
     
  12. JCLloyd

    JCLloyd New Member

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    Irving, from your "Cancelled/INVITE" string these extensions are repeated: 101, 102 & 110. I'd look at that ring group. My guess is that there are 4 extensions in that group. The missing extension would be my focus. What happens if you call that one?

    I have only a few months in the 3CX, myself, but this issue seems similar to one of the stumbling areas I went through.

    Keep at it. The more I use 3CX, the better it seems to be!

     
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  13. Irving

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    No only 3 extensions in the ring group. 101, 102, 110... all three have DID's if that matters.
     
  14. leejor

    leejor Well-Known Member

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    Well, "no audio" is generally a port issue. Either audio ports being blocked, or misdirected, network/firewall related. How do dialled DID numbers route? Do they route out on the trunk group, then back in through your provider, or do you use a loop trunk, or DIDs set as the SIP ID of the extensions.?
     
  15. JCLloyd

    JCLloyd New Member

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    Irving, I'm still learning 3CX, but this one is challenging me! I had the "no audio" condition when the IVR Server service was giving me some trouble. I'd bounce the SIP Server service, which bounced IVR & Media Server services, and it would come back. But, back to the 25% of calls not being picked up...
    . I was having a slightly different issue where I'd have to bounce the SIP Server service to get the 3CX to pick up inbound calls... Outbound was OK. Doing that bounces the IVR Server and Media Server services, and that would also take care of a dead audio condition which I experienced a couple of times.
    . I had another thought, which I will leave in at the end (see ***), but I am now wondering if you are having a SIP issue. I may be way off base, but you might see if your SIP provider can do a capture while you test. 3CX will want you to run Wireshark on the 3CX, which might catch something in the traffic. With only the 3 extensions, something external to the PBX might be causing the inbound connection to terminate.

    Please, put a note in with what you happen to find! I am not savvy enough to get much deeper on this one... yet...

    *** - Original thought...
    How are the extensions and ring groups set up for a 'no answer' situation, and do you have the ring group ringing longer than the individual extensions? Do you have the DID assigned in the extension, or as a separate inbound rule?

    I set up the two DIDs we currently have as inbound rules, so I could assign hours to them that do not match the main company hours as well as special routing for a no-answer. These are exceptions, and I would normally set the DID in the user extension. I am not sure if going one way over another would make a difference.
     
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  16. leejor

    leejor Well-Known Member

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    Individual forwarding options will be ignored for members of a ring group, unless, a call is direct to one extension, otherwise calls would be going off to members voicemails if the ringing timeouts had a conflict.
     
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  17. YiannisH_3CX

    YiannisH_3CX Support Team
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    Try enabling the option PBX delivers audio under extension settings and see it that resolves your audio issue. Now regarding calls going to voicemail you might need to run a wireshark capture on the server to be able to identify the issue. That should tell you why the call is being forwarded to voicemail.
     
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