Dismiss Notice
We would like to remind you that we’re updating our login process for all 3CX forums whereby you will be able to login with the same credentials you use for the Partner or Customer Portal. Click here to read more.

SIP Trunk

Discussion in '3CX Phone System - General' started by su27, Feb 1, 2010.

Thread Status:
Not open for further replies.
  1. su27

    Joined:
    Feb 1, 2010
    Messages:
    10
    Likes Received:
    0
    Hi ! This is my first install and first problems

    I'm trying to set up a sip trunk with a provider called COMS.com. I know it's not supported by 3cx but still believe I should be able to make it work

    I have two problems:

    1. when dial my VoIP number from my mobile - the call comes in, I pick it up - silence. The funny thing - my mobile doesn't realise that the call was answered - I still get ringing tone

    here is the log

    18:33:56.462 Looking for inbound target: called=02071111243; caller=02073883050
    18:33:56.460 [CM500002]: Info on incoming INVITE:
    INVITE sip:02071111243@192.168.5.11:5060;rinstance=8e2abad9f018aae8 SIP/2.0
    Via: SIP/2.0/UDP 85.90.225.100;branch=z9hG4bK0f33.141c031d845501c4aa9ec4b0d1f139af.0
    Via: SIP/2.0/UDP 85.90.225.100:5061;branch=z9hG4bKfc50344d489c73f717ef78cff502e2ad;rport=5061
    Max-Forwards: 16
    Record-Route: <sip:85.90.225.100;ftag=375981e35ecb7d3759075654fc276836o;lr>
    Contact: "Anonymous"<sip:85.90.225.100:5061>
    To: <sip:02071111243@85.90.225.100>
    From: <sip:02073883050@85.90.225.100>;tag=375981e35ecb7d3759075654fc276836o
    Call-ID: 7597452-3474038035-882516@MSX8.gammatelecom.com
    CSeq: 200 INVITE
    Expires: 300
    User-Agent: Sippy
    Content-Length: 0
    cisco-GUID: 570762408-2141142455-618591599-2729389085
    h323-conf-id: 570762408-2141142455-618591599-2729389085
    Portabilling-notify: aor=5101402835


    second problem - when i dial out, the call ends instantly

    18:24:15.810 [MS105000] C:3.1: No RTP packets were received:remoteAddr=85.90.225.100:42104,extAddr=195.137.29.160:32350,localAddr=195.137.29.160:32350
    18:24:08.902 [CM503008]: Call(5): Call is terminated
    18:24:08.849 [CM503020]: Normal call termination. Reason: Server Failure
    18:24:08.849 [CM503016]: Call(5): Attempt to reach <sip:02071149834@192.168.5.11> failed. Reason: Server Failure
    18:24:08.848 [CM503003]: Call(5): Call to sip:02071149834@sip.coms.com:5060 has failed; Cause: 500 Server internal error (TM); from IP:85.90.225.100:5060
    18:24:08.712 [CM503025]: Call(5): Calling VoIPline:02071149834@(Ln.10000@COMS)@[Dev:sip:5101402835@sip.coms.com:5060]
    18:24:08.708 [MS210002] C:5.2:Offer provided. Connection(transcoding mode): 195.17.129.160:32362(32363)
    18:24:08.502 [CM503004]: Call(5): Route 1: VoIPline:02071149834@(Ln.10000@COMS)@[Dev:sip:10140283@sip.coms.com:5060]
    18:24:08.500 [CM503010]: Making route(s) to <sip:02071149834@192.168.5.11>
    18:24:08.500 [MS210000] C:5.1:Offer received. RTP connection: 192.168.5.14:16400(16401)
    18:24:08.498 Remote SDP is set for legC:5.1
    18:24:08.497 [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:100@192.168.5.11:5060]
    18:24:08.496 [CM503001]: Call(5): Incoming call from Ext.100 to <sip:02071149834@192.168.5.11>
    18:24:08.493 [CM500002]: Info on incoming INVITE:
    INVITE sip:02071149834@192.168.5.11 SIP/2.0
    Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK-eeaff3d0
    Max-Forwards: 70
    Contact: "100"<sip:100@192.168.5.14:5060>
    To: <sip:02071149834@192.168.5.11>
    From: "100"<sip:100@192.168.5.11>;tag=d5f65b6b4dcd37bfo0
    Call-ID: 682a75db-c9aa600f@192.168.5.14
    CSeq: 102 INVITE
    Expires: 240
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Proxy-Authorization: Digest username="100",realm="3CXPhoneSystem",nonce="414d535c0177ad4812:81a656e27479e2aa96723b687e271f08",uri="sip:02071149834@192.168.5.11",algorithm=MD5,response="befcaf9231ab0b3531b089ab35e9762b"
    User-Agent: Linksys/SPA921-4.1.15
    Content-Length: 0


    am I missing something obvious?
     
  2. KerryG

    KerryG Active Member

    Joined:
    Jun 19, 2009
    Messages:
    960
    Likes Received:
    0
    Since it is an unsupported provider, did you try calling the provider for tech support? Maybe they know some 3CX settings.
     
    Stop hovering to collapse... Click to collapse... Hover to expand... Click to expand...
  3. turnkey

    Joined:
    May 9, 2009
    Messages:
    64
    Likes Received:
    0
    Hi,

    It could also be a firewall issue, have you ensured that you have all the relevant ports open on your router?
     
  4. su27

    Joined:
    Feb 1, 2010
    Messages:
    10
    Likes Received:
    0
    Yes, I tried contacting my Voip provider, but it's a "catch 22" situation - I have a voice provider that doesn't support 3CX PBX and I have a PBX that doesn't support my VoIP provider - so I'm on my own.

    I don't think it's a firewall issue - if I configure an individual phone to work with that voip provider - it works properly, so it must be 3CX that needs tweaking
     
  5. su27

    Joined:
    Feb 1, 2010
    Messages:
    10
    Likes Received:
    0
    posting my firewall tests

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

    <13:15:41>: Phase 1, checking servers connection, please wait...
    <13:15:41>: Stun Checker service is reachable. Phase 1 check passed.
    <13:15:41>: Phase 2a, Check Port Forwarding to UDP SIP port, please wait...
    <13:15:50>: UDP SIP Port is set to 5060. Response received correctly with no translation. Phase 2a check passed.

    <13:15:50>: Phase 2b. Check Port Forwarding to TCP SIP port, please wait...
    <13:15:59>: TCP SIP Port is set to 5060. Response received correctly with no translation. Phase 2b check passed.

    <13:15:59>: Phase 3. Check Port Forwarding to TCP Tunnel port, please wait...
    <13:16:08>: TCP TUNNEL Port is set to 5090. Response received correctly with no translation. Phase 3 check passed.

    <13:16:08>: Phase 4. Check Port Forwarding to RTP external port range, please wait...
    <13:16:18>: UDP RTP Port 9000. Response received correctly with no translation. Phase 4-01 check passed.
    <13:16:27>: UDP RTP Port 9001. Response received correctly with no translation. Phase 4-02 check passed.
    <13:16:36>: UDP RTP Port 9002. Response received correctly with no translation. Phase 4-03 check passed.
    <13:16:45>: UDP RTP Port 9003. Response received correctly with no translation. Phase 4-04 check passed.
    <13:16:54>: UDP RTP Port 9004. Response received correctly with no translation. Phase 4-05 check passed.
    <13:17:03>: UDP RTP Port 9005. Response received correctly with no translation. Phase 4-06 check passed.
    <13:17:12>: UDP RTP Port 9006. Response received correctly with no translation. Phase 4-07 check passed.
    <13:17:21>: UDP RTP Port 9007. Response received correctly with no translation. Phase 4-08 check passed.
    <13:17:31>: UDP RTP Port 9008. Response received correctly with no translation. Phase 4-09 check passed.
    <13:17:40>: UDP RTP Port 9009. Response received correctly with no translation. Phase 4-10 check passed.
    <13:17:49>: UDP RTP Port 9010. Response received correctly with no translation. Phase 4-11 check passed.
    <13:17:58>: UDP RTP Port 9011. Response received correctly with no translation. Phase 4-12 check passed.
    <13:18:07>: UDP RTP Port 9012. Response received correctly with no translation. Phase 4-13 check passed.
    <13:18:16>: UDP RTP Port 9013. Response received correctly with no translation. Phase 4-14 check passed.
    <13:18:25>: UDP RTP Port 9014. Response received correctly with no translation. Phase 4-15 check passed.
    <13:18:34>: UDP RTP Port 9015. Response received correctly with no translation. Phase 4-16 check passed.


    Application exit code is 0
     
  6. su27

    Joined:
    Feb 1, 2010
    Messages:
    10
    Likes Received:
    0
    update...

    According to my VoIP provider COMS.com - 3CX PBX floods the SIP trunk with "Status 200 OK" messages which it shouldn't do.

    they asked me to install a tool called Wireshark and make a phone call. The tool generated a log file, see attached. Whcih showed about 5 consequtive OK messages. According to them, these messages confuse their platform and it ends the call.


    but the question now is - how do I control 3CX so that it behaves? I can't seem to find any option to limit OK messages.
     

    Attached Files:

    • log.txt
      File size:
      5.5 KB
      Views:
      80
  7. TwigsUSAN

    Joined:
    Jan 31, 2007
    Messages:
    45
    Likes Received:
    0
    The reason why your VOIP provider is getting all the 200 oks is because they are not ACK the ok.
     
  8. mhanson

    mhanson New Member

    Joined:
    Nov 6, 2008
    Messages:
    186
    Likes Received:
    1

    Indeed. If they dont respond it is going to keep sending.
     
    Stop hovering to collapse... Click to collapse... Hover to expand... Click to expand...
  9. auto_emocion

    Joined:
    Feb 3, 2010
    Messages:
    7
    Likes Received:
    0
    Try opening All UDP ports on the firewall allowing into your 3CX. check firewall logs on the traffic and tighten up the rules.
     
  10. nb

    nb Support Team
    Staff Member 3CX Support

    Joined:
    Jun 7, 2007
    Messages:
    2,153
    Likes Received:
    170
    Try this:

    Disable Stun from the Advanced / network / STUN server section
    Enter Your public IP address in the requested box and select the correct outgoing interface.
    You will be prompted to restart the services and then try and make a call again.

    Let me know how this goes.
     
    Stop hovering to collapse... Click to collapse... Hover to expand... Click to expand...
Thread Status:
Not open for further replies.