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Solved Inbound Calls not working

Discussion in '3CX Phone System - General' started by Maarten, Aug 9, 2017.

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  1. Maarten

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    Hi there,

    I am completely new to 3CX and testing it to replace our current system.
    I have set everything up as i think is correctly. Trunk, inbound and outbound rules, extensions.
    Trunk is registering ok.
    I can call outbound with the extension fine, but inbound is somehow not working properly. The call is being received on the softphone. I see a brief window to accept the call for a second and it then closes and i get the message on the phone:

    "Cant get your call right now, please leave a message".

    The softphone shows that there is a missed call, so i presume it is working but somehow it's not going through properly.
    I ran the firewall checker and everything is in the green.
    Log file on the PBX shows this:

    08/09/2017 12:32:13 PM - [CM503003]: Call(C:13): Call to <sip:0001@txxxxxxxxxxxxx> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5080
    08/09/2017 12:32:02 PM - Discarded message, because of blocked User-Agent: BadUA Recv Req REGISTER from 217.182.205.85:5081 tid=20855cb0d7ac7c17598aba5b Call-ID=cb0d7ac-4ee57c17-598aba5b@217.182.84.84: REGISTER sip:217.182.84.84:5060 SIP/2.0 Via: SIP/2.0/UDP 217.182.205.85:5081;branch=z9hG4bK20855cb0d7ac7c17598aba5b;rport=5081 Max-Forwards: 70 Contact: "1670" <sip:1670@217.182.205.85:5081> To: "1670" <sip:1670@217.182.84.84:5060> From: "1670" <sip:1670@217.182.84.84:5060>;tag=20855cb153c3 Call-ID: cb0d7ac-4ee57c17-598aba5b@217.182.84.84 CSeq: 1 REGISTER Expires: 1800 User-Agent: VaxSIPUserAgent/3.1 Content-Length: 0

    If anyone could help or share some insight, that would be much appreciated.
     
  2. leejor

    leejor Well-Known Member

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    It appears that there may be a problem with the set-up of extension 1670. Is there only one instance of this extension on the PBX? If you change the routing to another extension, does it work correctly?
     
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  3. Maarten

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    This is strange as i have 1 extension on that trunk which is 0001. I can't find any 1670 extension in the pbx system.
    The trunk is of course linked to this 0001 extension.

    Tried one more time just now calling the trunk and i got this activity log:

    08/09/2017 3:00:15 PM - [CM102001]: Authentication failed for AuthFail Recv Req REGISTER from 91.134.110.224:55196 tid=1880522576 Call-ID=1808342562-219761609-328844064: REGISTER sip:217.182.84.84 SIP/2.0 Via: SIP/2.0/UDP 91.134.110.224:55196;branch=z9hG4bK1880522576 Max-Forwards: 70 Contact: <sip:237@91.134.110.224:55196> To: <sip:237@217.182.84.84> From: <sip:237@217.182.84.84>;tag=527351974 Call-ID: 1808342562-219761609-328844064 CSeq: 2 REGISTER Proxy-Authorization: Digest username="237",uri="sip:217.182.84.84",algorithm=MD5,realm="3CXPhoneSystem",nonce="414d5359598af94f91:4e9e59892f9a4f26e21add73dc363a72",response="51ebf138b67aa61ab53f3c5c97394288" User-Agent: sdfsfsdfsdfdf Content-Length: 0 ; Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings

    Not sure where this username: 237 comes from? or User-Agent: sdfsfsdfsdfdf ??
     
  4. Maarten

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    Also getting this error in the log:

    08/09/2017 3:54:50 PM - NAT/ALG check:L:10.1[Line:10000<<Withheld] REQUEST 'INVITE' - some of SIP/SDP headers may contain inconsistent information or modified by intermediate hop SIP proxy detected: Via:SIP/2.0/UDP 85.17.216.168;branch=z9hG4bKa43d.d083b92ca28720978b1907d7607908a0.0 Via:SIP/2.0/UDP 85.17.216.167:5060;received=85.17.216.167;branch=z9hG4bK45cdea03;rport=5060 Media session IP ('c=' attribute) is not equal to the SIP packet source(IP:port): Media session IP: 85.17.216.167 Received from: 85.17.216.168

    If this might be helpful
     
  5. lneblett

    lneblett Well-Known Member

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    A few things:

    It would help to know the extension number where the incoming call is destined; otherwise we are guessing.
    The 1670 is likely someone who is trying to hack into the system and make calls...given the user agent of VaxSIPUserAgent. It happens frequently and if you have a firewall that is robust enough to allow desired IPs and can then drop others, you should set this up.

    If extension 237 is the one where the inbound call is destined, then:
    <sip:237@217.182.84.84> From: <sip:237@217.182.84.84>;tag=527351974 Call-ID: 1808342562-219761609-328844064 CSeq: 2 REGISTER Proxy-Authorization: Digest username="237",uri="sip:217.182.84.84",algorithm=MD5,realm="3CXPhoneSystem",nonce="414d5359598af94f91:4e9e59892f9a4f26e21add73dc363a72",response="51ebf138b67aa61ab53f3c5c97394288" User-Agent: sdfsfsdfsdfdf Content-Length: 0 ; Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings

    I am not sure that the above 237 is the correct message as well as it is using a public IP and does not appear as registered and therefore should not be able to call out or receive any calls....even a flash of one.Having said that, it seems as though perhaps none of the clips you posted are relevant to the calls or extension of concern.
     
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  6. leejor

    leejor Well-Known Member

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    Are you using a hosted service, or is the server and the extensions all on the same LAN. What type of router are you using at the server end? The fact that the log indicates port numbers are "changing", is a concern and may indicate that a setting should to be changed.
     
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  7. Maarten

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    The PBX is hosted in the cloud. The router on the server side i am not sure? If you mean the router i am behind right now, that is a Microtik RB750. I have forwarded ports like 5060. opened up SIP ALG on the router if that is neccessary.

    The extension i am testing with is extension 2000. There is only 1 extension besides the default 0000. 1 Trunk which is linked to 2000. Outbound is working fine, but inbound doesn't go through. i see a brief window on the softphone of picking up the call which makes me assume that the call does go through somehow. It is less than a second of the window showing though and then it jumps to missed call and the message appears.
    We are using Nomado as our SIP trunk provider with lines etc.

    Is this because of the free edition?
     
  8. Maarten

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    This is the most current error i am getting which makes me suspect it's something with the network:

    08/10/2017 9:57:05 AM - [CM503003]: Call(C:25): Call to <sip:2000@xxxxxxxxxxxxxx:5060> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5080
    08/10/2017 9:57:05 AM - NAT/ALG check:L:25.1[Line:10000<<Withheld] REQUEST 'INVITE' - some of SIP/SDP headers may contain inconsistent information or modified by intermediate hop SIP proxy detected: Via:SIP/2.0/UDP 85.17.216.168;branch=z9hG4bK7163.ca19e6cb0c6201d22abf026e4ceb9cd6.0 Via:SIP/2.0/UDP 85.17.216.167:5060;received=85.17.216.167;branch=z9hG4bK41e1867e;rport=5060 Media session IP ('c=' attribute) is not equal to the SIP packet source(IP:port): Media session IP: 85.17.216.167 Received from: 85.17.216.168

    The extension up there is correct. Also, i am calling the trunk from a phone/number which is in the same lan as where the softphone is.
    Still, if i call with my own mobile for example, it shows the same result but instead of a 127.0.0 IP it shows a different one.
     
  9. Maarten

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    I found the problem. It seems the inbound calls were cancelled due to our other call center system having the same trunk registered as well. So the trunk was registered on two different spots. Once i removed the other one, calls were going through fine!
     
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