TIPS Can't get SIP trunk to register on port 5080 with IncrediblePBX13-12.2

Discussion in 'Open Discussion' started by tine, Feb 12, 2018.

  1. tine

    tine Member

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    Hi all
    I recently set up a Centos 6.9 box to run IncrediblePBX13-12.2. For most part it is working great, except that I can't get a SIP trunk with Voip.ms to register using port 5080. The box is connected behind a router. I tried port forwarding, but that did not help so I removed it. Here is what I have tried:
    1. Put Bindport=(blank)
    2. Added this to PBX firewall: -A INPUT -s 162.254.144.173 -p udp -m multiport --dports 5060,5080.4569,42872 -j ACCEPT
    3. Added .................................................:5080 to registration string.
    4. Added port=5080 to Peer Details for that trunk.

    Now if I setup an ATA to connect to the same provider's IP address on port 5080 it registers and works. This is showing that the problem is coming from the PBX.

    Eventhough I have the PBX set up to register on 5080, when I look at the registration status in my Voip.ms account, it shows the trunk registered on port 5060. The reason I need to change the port is because I have another trunk with another provider that is using 5060 on the same PBX. Any ideas?
     
  2. wardmundy

    wardmundy Nerd Uno

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    You can have more than one provider registered on port 5060. Have you tried disabling the firewall to see if that's the problem?
     
  3. tine

    tine Member

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    When I disable the firewall all my SIP Peer go down. IAX remain up.
     
  4. tine

    tine Member

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    Restarted the firewall, but SIP Peers still did not come up, so I rebooted the machine. All my SIP Peers are showing unreachable in PEER status on PBX.
     
  5. tine

    tine Member

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    My Voip.ms SIP trunk is now showing: "No registration found" in the Staus on my account.
     
  6. tine

    tine Member

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    A second iptables restart made my SIP Peers come up, but I still have the original problem. The issue is that when I call the DID assigned to this trunk, you get a busy. I think that is because another trunk is already using 5060 on the same Public IP. I can't change the other trunk, the provider only uses 5060.
     
  7. islandtech

    islandtech Wassamassaw

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    Try using ip authentication
     
  8. tine

    tine Member

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    Does IP Authentication work when PBX is behind a router? If so, how would the inbound work?
     
  9. tine

    tine Member

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    Actually the problem is not that the trunk is not registering, it is. I can make an outbound call. Inbound get a busy because i guess the port is being used by another trunk.
     
  10. wardmundy

    wardmundy Nerd Uno

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    You've indicated that the trunk is not registering and then you said the trunk was registered. Which is it?? sip show registry will tell you on the Asterisk CLI. Assign ALL of your trunks to UDP 5060 until you get things working. Your problem is NOT with multiple providers sending calls over the same port. You can have dozens of trunks registered and sending traffic on UDP 5060. If the inbound call is failing, take a look at the Asterisk CLI and see if the call is even reaching your PBX. If so, what is the reported problem? Do you have an inbound route specified for the DID?
     
  11. billsimon

    billsimon Experienced in Asterisk, FreePBX, and SIP

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    Stop guessing; look at the logs (/var/log/asterisk/full) to see what the actual problem is. Registration isn't for outbound calls, it's for receiving calls. Always troubleshoot the two call directions separately.
     
  12. mainenotarynet

    mainenotarynet Not really a Guru - Just a long time user

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    There should be One inbound route already named Default with a CID on 2nd line of Any/Any. See what this route has for what it does with calls (I think it just Hangs Up, unless it was changed [I set mine to put caller on hold forever withe a MOH class of a streaming audio source, So I can listen to music by dialing 7777 {Simulate incoming call}])

    That is another way to find where the problem is call 7777 and see if YOU get the busy signal. I have voip.ms and rarely (and over 6+ years I mean RARELY) have problems. I do not use IP Auth - I have 12 character passwords -- I can help to see where the break is Check your Voip.ms account and see if the trunks are registering on their end (bottom of portal page) if they are then the calls are hitting the PBX (or should be anyways) if not we need to work back from voip.ms and check the trunk configs (I have working ones)

    If you can't get this going find my # on my U/N.org contact page and we can work on the issue by voice and post back when fixed. Sometimes a 5 minute call can save days of forum posts :smile5:
     
  13. tine

    tine Member

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    If you guys read my original post you will see that I said that everything was working except for inbound call on one trunk. I also mentioned that in my pbx config I have my Peer details and registration string set for port 5080 yet in my VOIP.ms account page it is showing up as registering as port 5060, which I believe is causing me issues. The real question is: Can IncrediblePBX be made to register a SIP trunk on another port other that 5060? I have proven that the problem is on the PBX. I have 5 trunks connected to this box, 3 different providers. One IAX2 and 4 SIP All I want is that each trunk is on a different port. I should be able to accomplish this with this amount of trunks, and also eliminate the problem I mentioned in the OP. Thanks
     
  14. billsimon

    billsimon Experienced in Asterisk, FreePBX, and SIP

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    Yes.

    What it's showing you is that you are registered to receive calls on YOUR port 5060. That doesn't mean you're not communicating with voip.ms on THEIR port 5080. Do "sip show peers" and "sip show registry" on the Asterisk CLI to see whether you are connecting to voip.ms port 5080 or not.
     
  15. tine

    tine Member

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    Both "sip show peers " and "sip show registry" shows the trunk registering on port 5080, but this is what I have been saying. Why doesn't my VOIP.ms account show the same as it shows my IAX2 trunk and my SIP trunk connected to an ATA?
     
  16. billsimon

    billsimon Experienced in Asterisk, FreePBX, and SIP

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    When you set up your trunk you are telling Asterisk to connect to VoIP.ms on port 5080. It registers and says it can receive calls at its own IP and port (bindport). In your case that is 5060. This is how it works. Now that I understand what you are trying to do, which is have Asterisk listen for sip on multiple udp ports, the answer is no. You can't do that. Did you work with FreeSWITCH at some point? FS works like that and they are called "SIP profiles" on that platform.
     
  17. tine

    tine Member

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    Thanks for the responses.

    Bill,
    Your last response was a bit confusing. My "bindport" setting is blank. Unless extensions work differently, I have all my extensions running on different ports. Wasn't necessary for some but I did it anyway. So you are saying that a SIP trunk can only register on 5060. I am sure that I had a previous version of Asterisk( PIAF or Incredible) that showed my registration as 5080 in my VOIP.ms account.

    Thanks again for your input.
     
  18. billsimon

    billsimon Experienced in Asterisk, FreePBX, and SIP

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    On extensions you can easily set the port it listens on as well as the port it connects to. So it sounds like you are setting different listen ports on all your extensions (but why?). That same setting is "bindport" in Asterisk but because it acts as a server and a client (it is a server to your extensions, a client to your trunks) you probably want to keep it on 5060.
     
  19. tine

    tine Member

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    Hi Biill,
    Since my "bindport" is blank, where did the 5060 come from. I also noted that it was Asterisk that recommended that it be left blank.

    I read that I can have one trunk to my provider and use Inbound routes?
     
  20. tine

    tine Member

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    I changed my setup to now have one IAX2 trunk to Voip.ms and then route the DIDs on that trunk. Used my same inbound route. It is now working as expected. Thanks again for all you input.
     

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