Sipgate - Busy tone on incoming

Discussion in '3CX Phone System - General' started by daiablo, Apr 23, 2009.

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  1. daiablo

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    Hello everyone,

    I have set up 3cx on one of our local Windows 2008 servers here and configured our Draytek 2800VG router firewall to pass all the relevant ports through to the server. I have set up a sipgate account with a local geographic telephone number and created the relevant VOIP provider entry in the 3cx system and this shows as registered. I have set up and outbound rule that anything that starts with 9 goes out via the Sipgate provider which appeas to work fine - I get the message saying there is no credt for outgoing calls, etc... I have created an inbound rule that all calls go to extension 100 which is a registered Linksys desk phone. The extension is set to use PBX status for busy detection. I have installed the soft phone onto my laptop and can make internal calls between the soft phone and the desk phone without problems.

    The problem comes when I try to call the geographic sipgate number. All I get is a busy tone and there is no record of the incoming call in the server activity log even when set to verbose. This leads me to believe that the problem may be something in my firewall setup. I have re-checked the Draytek configuration and also the configuration of the Windows Server 208 firewll on the machine and they appear to be correct. I have run the firewall checker and it passes, with some warnings.

    Does anyone have any ideas where I may be going wrong? Or some other steps for me to try? Any feedback is welcome - feel free to call me a noob at anytime!

    Thanks,
    Daiablo
     
  2. leejor

    leejor Well-Known Member

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    Since your outgoing calls are greeted with a recording saying that you have no credit, i'm wondering if you need credit to receive incoming calls? Perhaps try setting up a number with ipkall.com, it's free. You need to have a fixed public IP address or be using a DDNS service to track your IP.
     
  3. daiablo

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    Thanks for the response leejor. According to the Sipgatewebsite this isn't the case but you never know!

    I will give ipkall.cm a go an see how I get on. thanks for the advice.

    Regards,
    Daiablo
     
  4. daiablo

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    Have confirmed that Sipgate account doesn't need to be topped up to receive incoming calls.

    Unfortuantely IPkall.com is a washington based number and I am based in the UK. If I use this to test the system and call the number from our office PSTN line then my boss is going to go mad when the phone bill comes through :) I am unable to find another UK equivelant to test.

    I can't help thinking that this is some sort of firewall or registration problem for the call not to come through. Does anyone else have some other advice.
     
  5. leejor

    leejor Well-Known Member

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    I'm asumming that you forwarded all the ports( 5060,9000->9049,5090,5080) to the computer running 3CX and that it has a fixed IP? What were the warnings when you did the firewall check?
     
  6. daiablo

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    Here is the output of the firewall checker


    ----------------------
    3CX Firewall Checker, v1.0. Copyright (C) 3CX Ltd. All rights reserved.

    <09:13:03>: Phase 1, checking servers connection, please wait...
    <09:13:03>: Stun Checker service is reachable. Phase 1 check passed.
    <09:13:03>: Phase 2a, Check Port Forwarding to UDP SIP port, please wait...
    <09:13:08>: UDP SIP Port is set to 5060. Response received WITH TRANSLATION 37924::5060. Phase 2a check passed with WARNINGS. Some functionality will be LIMITED. For more information, please visit https://www.3cx.com/blog/docs/firewall-checker/

    <09:13:08>: Phase 2b. Check Port Forwarding to TCP SIP port, please wait...
    <09:13:12>: TCP SIP Port is set to 5060. Response received WITH TRANSLATION 37924::5060. Phase 2b check passed with WARNINGS. Some functionality will be LIMITED. For more information, please visit https://www.3cx.com/blog/docs/firewall-checker/

    <09:13:12>: Phase 3. Check Port Forwarding to TCP Tunnel port, please wait...
    <09:13:17>: TCP TUNNEL Port is set to 5090. Response received WITH TRANSLATION 37954::5090. Phase 3 check passed with WARNINGS. Some functionality will be LIMITED. For more information, please visit https://www.3cx.com/blog/docs/firewall-checker/

    <09:13:17>: Phase 4. Check Port Forwarding to RTP external port range, please wait...
    <09:13:23>: UDP RTP Port 9000. Response received WITH TRANSLATION 41864::9000. Phase 4-01 check passed with WARNINGS. Some functionality may be IMPAIRED. For more information, please visit https://www.3cx.com/blog/docs/firewall-checker/
    <09:13:27>: UDP RTP Port 9001. Response received WITH TRANSLATION 41865::9001. Phase 4-02 check passed with WARNINGS. Some functionality may be IMPAIRED. For more information, please visit https://www.3cx.com/blog/docs/firewall-checker/
    <09:13:32>: UDP RTP Port 9002. Response received WITH TRANSLATION 41866::9002. Phase 4-03 check passed with WARNINGS. Some functionality may be IMPAIRED. For more information, please visit https://www.3cx.com/blog/docs/firewall-checker/
    <09:13:36>: UDP RTP Port 9003. Response received WITH TRANSLATION 41867::9003. Phase 4-04 check passed with WARNINGS. Some functionality may be IMPAIRED. For more information, please visit https://www.3cx.com/blog/docs/firewall-checker/
    <09:13:41>: UDP RTP Port 9004. Response received WITH TRANSLATION 41868::9004. Phase 4-05 check passed with WARNINGS. Some functionality may be IMPAIRED. For more information, please visit https://www.3cx.com/blog/docs/firewall-checker/
    <09:13:45>: UDP RTP Port 9005. Response received WITH TRANSLATION 41869::9005. Phase 4-06 check passed with WARNINGS. Some functionality may be IMPAIRED. For more information, please visit https://www.3cx.com/blog/docs/firewall-checker/
    <09:13:50>: UDP RTP Port 9006. Response received WITH TRANSLATION 41870::9006. Phase 4-07 check passed with WARNINGS. Some functionality may be IMPAIRED. For more information, please visit https://www.3cx.com/blog/docs/firewall-checker/
    <09:13:54>: UDP RTP Port 9007. Response received WITH TRANSLATION 41871::9007. Phase 4-08 check passed with WARNINGS. Some functionality may be IMPAIRED. For more information, please visit https://www.3cx.com/blog/docs/firewall-checker/

    Application exit code is 11
    ------------------------------

    My guess is something to do with the RTP ports being impaired but I don't know where to look to resolve this. All the ports listed are being forwarded from the external IP address to the internal static IP Address.

    thanks,
    Daiablo
     
  7. daiablo

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    As an additional step I have added the internal server as a DMZ host on my router and also turned off Server 2008 firewall and run the checker again and it comes up with the same results.

    Thanks,
    Daiablo
     
  8. leejor

    leejor Well-Known Member

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    It looks like the STUN test is having some sort of problem. What type (make) of router are you using? Do you have a second one, different make, to test with? You could try putting the server right on the "net" so that it gets a public IP, no need for STUN, and see if it works. You'd then know if it is a problem with the router. You don't have Zonealarm or any other firewall running?
     
  9. daiablo

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    We are using a Draytek 2800VG. Unfortunately all the other routers we have are also Drayteks so will most likely have similar issues. The only other firewall running is the windows firewall on the server 2008 machine - which I have disabled.
     
  10. swim

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    are you using the sipgate template built into 3cx?
    is the line registered on the 3cx server and in your sipgate account ?
    can you reach the sipgate test number ? dial 10000 from your registered phone

    This problem can sometimes be overcome by using STUN in conjunction with a proxy. The port sipgate uses for STUN is 10000, but the standard STUN port is 3748 (forward both). The proxy server for sipgate is sipgate.co.uk
     
  11. ciscotech2007

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    Maybe something is incorrectly configured somewhere. Our advice.

    note: this is not detailed process but a rough overview

    (1) make a backup of your system via 3CX backup utility
    (2) remove 3CX server
    (3) use 3CX installation checker to ensure no 'pieces' of 3CX are still on your machine
    (4) we may even recommend resintalling IIS or Cassani web servers (whichever your using)
    (5) reinstall your selected web server (iis *OR cass. NOT both)
    (6) reinstall 3CX server, verify no errors during install
    (7) after 3CX server install, auto-run 3CX inital wizard will start
    (8) for FQDN insert the IP ADDRESS of your 3CX server, which should be static IP (not DHCP)
    *you can use FQDN if your internal network has internal DNS server w/ SRV records for SIP
    (9) choose (3) digit extensions
    (10) add ONLY 2 extensions, one for your 3CX VoIP softphone and 1 of your SIP hardphones. If would be better to use (2) computers each with their own softphone but some people don't have (2) computers.
    (11) configure all IP endpoints with SIP registration information (i.e. extension, ID, password, etc.)
    (12) ensure all extensions register in your 3CX extension status page (i.e. green light)
    (13) call from your (1) internal extension to the other internal extension, should have success, if you don't there is an issue
    (14) if no success, change 3CX server logging to verbose and post the log in this forum post. Remember to go to the log right AFTER your phone call fails.
    (15) if your call completes to the other internal phone extension you have verified the 3CX system works on a basic level
    (16) register an account with a 3CX RECOMMENDED AND VERIFIED VoIP trunk provider, like Broadvox, CallCentric and more.
    (17) follow the instructions for adding a VoIP provider here:
    http://www.3cx.com/manual/3CXPhoneSyste ... tem37.html
    (18) make sure the Add VoIP provider wizard sets up an OUTBOUND rule
    (19) make an 'easy' OUTBOUND rule... select '9' to the be ID for OUTBOUND CALLS. Meaning all calls beginning with '9' are to be forwarded to your VoIP provider
    (20) make sure the OUTBOUND rule strips the first digit after dialing out.. which is your '9'
    (21) check your dialplans (if any) and 3CX server logs and CONFIRM the OUTBOUND dialed number is correct.. if you only see '234' or '23444' something it wrong in your dialplan or OUTBOUND prepend or strips
    (21) CONFIRM with your VoIP provider what digits you must dial during OUTBOUND calls.. meaning.. some voip providers REQUIRE you to dial your destination area code even while dialing local numbers. Some require you to include a country code as well. So if you have a 1-123-456-7899 number from your voip provider, you may have to dial 9+1+XXX+XXX+XXXX even on local calls to the same area code as your number provided by your voip provider, in this case, '123'.
     
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  12. daiablo

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    Hi Swim,

    Yes, I used the Sipgate template and the primary stun server is set to stun.sipgate.net and port 10000. The secondary stun server is the 3cx stun server with the usual port. The line registers fine and I can make outgoing calls including the sipgate test number. It's incoming calls I'm not receiving. I will try the proxy server and let you know how I get on.

    Thank,
    Daiablo
     
  13. daiablo

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    Tried it with a proxy but still no difference. Just a busy tone on all incoming calls even though at least one extension is free and there are no other active calls on the system. Thanks for the info though.

    Daiablo
     
  14. swim

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    Hi Daiablo,

    I dont think the problem is with 3cx, but with Draytek router.

    Try and see wether you can register one Linkysys phone directly with Sipgate and by pass 3cx for testing purposes.
    You can log into your Sipgate account and get all the required settings for your phone from > Help-Centre Devices - configuration and troubleshooting > Linksys

    You should also try to adjust your router/firewall to pass through voice packets (forwarding them). You will likely have to go into your router/firewall settings under port forwarding and forward the following ports:

    Standard:
    Port: 5060 / UDP (SIP signaling)
    Port: 5004 / UDP (RTP, language)
    Port: 10000 UDP (STUN)

    on Lynksys try stun setting of : stun.sipgate.net:10000 or stun.sipgate.net

    let me know how you get on

    Swim
     
  15. daiablo

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    Hi Swim,

    Changed the stun server and it registered correctly but the same results. All the ports you suggested are open and forwarding. I'm wondering if the fact that the draytek is a 2800VG (V standing for VOIP) it is stopping the packets coming through as it recognises them as VOIP/SIP packets and tries to use them. There are no VOIP accounts set up on the router so there's no reason hy it shouldn't forward the traffic accordingly.

    I have registered the phone directly with sipgate and I get the same results. Outgoing works fine but incoming is just a busy tone. Still behind the Draytek router/firewall.

    Thanks,
    Daiablo
     
  16. swim

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    Hi Daiablo,

    Draytek 2800VG should be fine, I have used a Draytek 2600VG in the past with no issues. I still think the issues are down to stun settings / draytek router set up. What security set up do you have in the router, are you blocking udp flood, I had issue with this setting turned on, try turning it off, failing this try e-mailing sipgate support they normally respond within 24hrs.

    Swim
     
  17. daiablo

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    Swim,

    Here's the open ports setup on the Draytek.
    "Image not found"

    Does that all look correct to you?

    I've turned off UDP flood but it's made no difference. Thanks for sticking with me on this - it's most appreciated!

    Thanks,
    Daiablo
     
  18. swim

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    Hi Daiablo,

    ok turn UDP flood back on and set ports to

    UDP start 9000 finish 9015
    TCP start 5060 finish 5060
    UDP start 5060 finish 5060
    TCP start 5090 finish 5090

    this will be used for 3cx or direct with sipgate

    UDP start 5004 finish 5004 used direct with Sipgate

    this will make the start and end ports the same


    Swim
     
  19. daiablo

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    Hi Swim,

    Made the changes as you suggested. I had them previously but read another post somewhere regarding SIP sometimes needing contiguous ports so changed the ranges. Should have changed them back when I realised I was barking up the wrong tree.

    Is it worth me trying a different sip provider to see if that works. Preferably someone who gives a free sip account with a UK geographical phone number.

    Thanks,
    Dave
     
  20. swim

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    Hi Dave

    I have sent you a PM

    Swim
     
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