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Inbound and outbound calls need to be made twice to hear

Discussion in '3CX Phone System - General' started by Paulh, Sep 11, 2008.

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  1. Paulh

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    I have a problem where I need to call an outbound number twice to get through. The first time I call, I get silence. If I call again, it works. Same with inbound calls. When the phone rings, I get silence. Then if the other person rings again, I can hear them. The other party hears silence too the first time. I keep getting voicemails with silence, so not sure if anyone's leaving messages or hanging up.

    It's extremely frustrating would like to resolve the issue rather than change phone servers.

    Phone is Linksys SPA941 or Snom M3 (happens with both - although the M3 doesn't ring half the time, but that's another issue). Using NexVortex as the IP phone provider.

    Does anyone know why this may be happening?

    Paul
     
  2. ecwilson

    ecwilson New Member

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    Hi Paulh

    We need to know a little more about your setup & some log files from the server.
    Also did you do a fire wall test on the server??? Here are a few other questions.
    1)...what type of computer is the 3cx running on
    2)...what type of router are you using
    3)...what type of switch are you using

    These are just some basic questions for others to help you with your problem.
    It also sound like something is opening & closing on your network (ports) for
    some reason the port does not open until the second call attempt :?:

    ecwilson
     
  3. Paulh

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    Hi ECwilson,

    Thanks for your response. Here's some answers:

    1. Core 2 Quad 9300, Intel DG31 motherboard, Sata drive(s), 2GB RAM
    2. Router is a Dlink DGL-4100
    3. As above

    Port 5060 is open. The firewall test passes.

    Here is the log. You can see the first call doesn't even go through. Looks like a firewall issue. But the second time it does. Why would it not work the first time, but work the second time. Also, sometimes it does work the first time, but not usually. (I've removed IP address, usernames and phone numbers from the log).


    ---------------------------------------
    09:31:08.990 Call::Terminate [CM503008]: Call(60): Call is terminated
    09:31:08.928 Call::Terminate [CM503008]: Call(60): Call is terminated
    09:30:43.819 CallCtrl::eek:nLegConnected [CM503007]: Call(60): Device joined: sip:<removed>
    09:30:43.803 CallCtrl::eek:nLegConnected [CM503007]: Call(60): Device joined: sip:100@192.168.1.167:5060
    09:30:37.491 Line::printEndpointInfo [CM505003]: Provider:[NexVortex] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.1.99:5060]
    09:30:37.491 CallCtrl::eek:nAnsweredCall [CM503002]: Call(60): Alerting sip:<removed>
    09:30:33.553 CallCtrl::eek:nSelectRouteReq [CM503004]: Call(60): Calling: VoIPline:<removed>@(Ln.10001@NexVortex)@[Dev:sip:<removed>:5060]
    09:30:33.553 CallCtrl::eek:nSelectRouteReq [CM503010]: Making route(s) to [sip:<removed>@192.168.1.99]
    09:30:33.553 Extension::printEndpointInfo [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA941-5.1.8] Transport: [sip:192.168.1.99:5060]
    09:30:33.506 CallCtrl::eek:nIncomingCall [CM503001]: Call(60): Incoming call from Ext.100 to [sip:<removed>@192.168.1.99]
    09:30:23.006 Call::Terminate [CM503008]: Call(59): Call is terminated
    09:30:22.975 Call::Terminate [CM503008]: Call(59): Call is terminated
    09:30:21.881 CallCtrl::eek:nLegConnected [CM503007]: Call(59): Device joined: sip:<removed>:5060
    09:30:21.866 CallCtrl::eek:nLegConnected [CM503007]: Call(59): Device joined: sip:100@192.168.1.167:5060
    09:29:58.491 Line::printEndpointInfo [CM505003]: Provider:[NexVortex] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.1.99:5060]
    09:29:58.491 CallCtrl::eek:nAnsweredCall [CM503002]: Call(59): Alerting sip:<removed>:5060
    09:29:54.928 MediaServerReporting::InitEndPoint [MS003005] C:59.2: Failed to create Endpoint: (destination=<removed>)
    EndPoint: ID=00000096@(EXTERNAL)
    LOGID=C:59.2 Status: MSEP_FAILED
    RTP:192.168.1.99:9006
    RTCP:192.168.1.99:9007
    STUN RTP:0.0.0.0:0
    STUN RTCP:0.0.0.0:0
    Coder:
    NOT SET
    101:telephony-event
    Party ptime:20
    Party RTP:0.0.0.0:0
    Party RTCP:0.0.0.0:0
    Decoders:
    [empty]

    09:29:54.928 MediaServerReporting::InitEndPoint [MS003002] C:59.2: RTP socket 192.168.1.99:9006 binding failed with error code 10048
    09:29:54.928 MediaServerReporting::STUN [MS101003] C:59.2: Possible firewall problem. Address mapping failed on STUN server 75.101.138.128:3478 for local address ":9006"
    09:29:54.850 CallCtrl::eek:nSelectRouteReq [CM503004]: Call(59): Calling: VoIPline:<removed>@(Ln.10001@NexVortex)@[Dev:sip:<removed>:5060]
    09:29:54.803 CallCtrl::eek:nSelectRouteReq [CM503010]: Making route(s) to [sip:<removed>@192.168.1.99]
    09:29:54.788 Extension::printEndpointInfo [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA941-5.1.8] Transport: [sip:192.168.1.99:5060]
    09:29:54.507 CallCtrl::eek:nIncomingCall [CM503001]: Call(59): Incoming call from Ext.100 to [sip:<removed>@192.168.1.99]
    -----------------------------

    Thanks again for any advice...

    Paul
     
  4. ecwilson

    ecwilson New Member

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    Paulh

    Problem One the stun server must not have any errors :?:

    09:29:54.928 MediaServerReporting::STUN [MS101003] C:59.2: Possible firewall problem. Address mapping failed on STUN server 75.101.138.128:3478 for local address ":9006"

    Also I will assume you are on WinXP of Vista :?:
    Is the 3cx box in the DMZ of the router :?:
    What does the phone say about the Port Type :?:
    How is the phone set for stun resolution :?:

    ecwilson
     
  5. Paulh

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    Ecwilson,

    The server is running on SBS2003.
    The server is not in a DMZ
    STUN is turned off

    Not sure what you mean by port type.

    Thanks again!

    Paul
     
  6. ecwilson

    ecwilson New Member

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    Paulh

    Without stun on how will nexvortex system find your system unless you have a public ip :?:
    Also stun resloution keeps ports open on the router so udp communications can travel both
    ways & when they are not opened you get one way audio or no audio first them audio second.
    Your problem is getting the nexvortex info to the server & then back out to NV. Is stun on or
    off on the 3cx server :?: PM me when you get a chance :)

    ecwilson
     
  7. AJAY

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    Sound like closed ports to me.

    Have you opened ports on your router as laid out in the manual

    Ports to be opened -
    5060 (UDP)
    900-9015 (UDP)
    5090 (TCP) for Tunnel if used.

    The firewall checker is pretty much on the money with it's reports and is useful for checking and rectifying non traversal
     
  8. Paulh

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    Hey Ajay,

    I have the ports you mentioned opened. (TCP: 5060, 5090; UDP: 5060, 9000-9015)

    Firewall checker is reporting no problems.

    Thanks
     
  9. ecwilson

    ecwilson New Member

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    Paulh

    If your 3cx box does not have a public ip you can put it in the dmz zone & that
    should bypass your port issue but you will still need to have stun active on the
    server. For now I can't find the post but there is a list of all the ports that need
    to be opened.

    3478
    5060
    7000-7400
    9000-9015

    ecwilson
     
  10. Paulh

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    I have a static IP. STUN is turned on in 3CX. I'm loathe to but the server in a DMZ because it's running on a SBS2003 server...
     
  11. ecwilson

    ecwilson New Member

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    Paulh

    I don't understand this line :?:

    I'm loathe to but the server in a DMZ because it's running on a SBS2003 server... :?:

    Explain :?: That statement :?:

    Can the ip address of the 3cx box be pinged from the outside :?:

    ecwilson
     
  12. Paulh

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    I don't want to put the 3CX machine in the DMZ because it's running on a SBS2003 server.

    I also have a Packet 8 phone and that's working without any problems.

    So I have all the ports open I need, I'm pretty sure everything is setup as it should be, but I'm still having this issue.

    At this stage, I think I need to give up on 3CX becuase I've already lost way too many calls.
     
  13. ecwilson

    ecwilson New Member

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    "SEE BELOW"

    by Paulh on Wed Sep 17, 2008 6:13 am

    I don't want to put the 3CX machine in the DMZ because it's running on a SBS2003 server.
    :| At this point I don't understand why you don't want to put it in the dmz SBS2003 is a data
    center level operating system & was designed to run in a data center where there are no
    nat setups just public ip's open to the world :?:

    I also have a Packet 8 phone and that's working without any problems.
    :| Packet8 is what we call a closed system like (vonage/skype) so they setup
    things very different than the rest of the voip world does :?: Also P8 someone
    else did all the setup that you now have to do with the 3cx :cry:

    So I have all the ports open I need, I'm pretty sure everything is setup as it should be, but I'm still having this issue.
    :| The 3cx is very easy to setup but your router is a different story so don't give up on the 3cx yet. We use it & it
    works just fine but we are also looking at a multi-tenent ip pbx to better fit our needs.

    At this stage, I think I need to give up on 3CX becuase I've already lost way too many calls.
    :| Don't give up yet :idea: Send me a pm so we can do a test with another 3cx box from
    your location :?:

    ecwilson
     
  14. Alejandro Esteve

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    Hi,

    i have a similar problem, in my case i can ring with my softphone thought a tunnel and it work´s fine but if rings my remote extension one time i can hear and de second time can´t hear, and it´s repeat more times.

    I probe with a geografic number, with an SPA3102 pstn line and in both cases, is the same. The firewall test is right.

    Haver you a posible solution for this ?

    Thnx.
     
  15. Alejandro Esteve

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    Hi to all,

    I find the problem of this issue. The problem is the SoftPhone client 3CX version 6.0.727.0 and Windows Vista Ultimate.

    It haven´t works properly. I test with the X-Lite client and works fine. Then, the problem isn´t the router or the nat or other networking paramenters.

    We hope 3cx solve this problem nearly as posible.

    Thanks.
     
  16. sharksfan

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    Check out my issue/resolution

    http://www.3cx.com/forums/random-no-audio-issue-v7-7821.html
     
  17. itglaafs

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    Dear all,

    Following this link does not open up another page...it seem like it is removed. Anyways, we are experiencing same issue, randomly audio is being dropped in the middle of the call, you have to call in twice in order to get through. We are using V15.5 with Pro license.

    I have just connected Eyebeam soft phone to 3CX, and first impressions are good. We were able to get the number we couldn't reach out in last couple of days with 3CX client.

    Any ideas or suggestions?

    Txs.
     
  18. OCWI

    OCWI New Member

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    This is definitly a firewall/NAT issue. Two variables here.

    1) Make sure SIP ALG is turned off on the ISP modem. This will cause this exact them. The ISP's around us lately have been hung up on the difference between bridge mode and straight bridge mode. You want straight bridge.

    2) If the issue remains, try hosting the PBX on OVH (its $3/mo...) and put the phone(s) behind a session border controller that is running on a raspberry pi 3.

    #1 has a chance of fixing.
    #2 WILL fix it
     
  19. itglaafs

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    Thank you for your reply.

    SIP ALG is turned off, I have verified it on firewall interface. Ports are being forwarded as per instructions, Firewall test says that it's all OK.

    We have static ip address and are on business fiber optic internet connection with symmetric download speed. So the quality of internet and ISP setup is not an issue.

    If I run Eyebeam or Bria from Counterpath with 3CX PBX, I don't have these issues, we tested it last night, with 8 hour run of calls. This only happens with 3CX client. I have disabled 3CX tunneling when out of the office features, so I will see if this solves the issue.

    Any other thought? Thank you very much.
     
  20. itglaafs

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    The issues have returned back, Eyebeam was just holding for few days, so we switched back to 3CX clients.

    I have disabled SIP ALG on router and verified that all ports that needs to reach PBX are being forwarded to it. 3CX PBX reports that all firewall tests are OK.

    This time I have skipped port forwarding and created manual rules that pass on necessary ports from WAN to the local IP address of 3CX. This has improved calls much more and now every fourthy or fifthy call gets one way audio. Of course when you redial, call is fine.

    I have uploaded Wireshark part of one such call, perhaps someone with more experience is able to see what's wrong with it.

    Thank you very much.
     

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