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GENERIC SIP Configuration

Discussion in '3CX Phone System - General' started by Trinexus, Aug 22, 2016.

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  1. Trinexus

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    Hi:

    I am new to 3CX and the communications arena. I have a 3CX installation with 3 Voip lines incoming from a service in US (we are located in Puerto Rico). The Voip Service is presenting some challenges due to latency. I request a SIP trunk from a local provider (Claro). Now I have to configure it.

    From all the information available, I understand that the SIP should be configured as a Generic SIP / IP Based - provider only gave me an IP and when I asked they told me they do not provide a login/authentication password for the lines.

    The SIP have 36 DIDs.

    I installed a second NIC to the server and connected the SIP directly to the Provider's equipment. I did configured the IP on the card. I did create the Generic SIP /IP Based entry on the 3cx. All the test I did (incoming/outgoing) are failing.

    I would like to start by defining that some specific calls are send through this new SIP to test it (outgoing).

    Can anyone guide me where to find the correct steps for a case like this? Should I create static routes on the server for a specific service and set something on the 3CX?

    Any help will be appreciated.
     
  2. leejor

    leejor Well-Known Member

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    Many users avoid two NICs as issues can arise. This does have some information about adding the second NIC. (it's a few years old)

    http://www.3cx.com/blog/docs/network-co ... ne-system/

    Not sure if your set-up will be similar. You might want to provide additional information about how (public/Private IP) the cards are configured. Someone else might have the same set-up.
     
  3. Trinexus

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    I did configured the 2 NICS s suggested. I was able to generate a call through the new SIP trunk but the recipient did not hear me, even though I can hear him loud and clear. The calls on my Voip Services also fails some times (the recipients cannot hear us), so I am wondering if this is a global setting issue or is at the SIP trunk definition. Any ideas?

    One thing I just notice are these entries in the Activity Log:
    08/29/2016 8:36:20 PM - NAT/ALG check:L:1136.2[Line:10003>>7873658813] RESPONSE 200 on 'INVITE' - some of SIP/SDP headers contain inconsistent information or modified by intermediate hop
    Media session IP ('c=' attribute) is not equal to the IP specified in contact header:
    Media session IP:66.50.186.185
    Contact IP:66.50.186.16
    Media session IP ('c=' attribute) is not equal to the SIP packet source(IP:port):
    Media session IP: 66.50.186.185
    Received from: 10.36.129.209

    08/29/2016 8:36:10 PM - NAT/ALG check:L:1136.1[Extn] REQUEST 'INVITE' - some of SIP/SDP headers may contain inconsistent information or modified by intermediate hop
    SIP contact header is not equal to the SIP packet source(IP:port):
    Contact address:192.168.100.126:57694
    Received from :192.168.100.126:63144

    Any ideas?
     
  4. leejor

    leejor Well-Known Member

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    Some routers do port substitution, which, while concerning, in many cases, seems to cause no real issues. The fact that you have mixed in two NICs may be compounding a problem.

    Have you run the 3CX Firewall Checker with success?
     
  5. Trinexus

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    Yes, I ran the firewall checker and everything is fine. One NIC have access to the internet (connected to the internal router and the firewall), providing access to the Voips, the other NIC is connected directly to the SIP Provider's router.
    I had assigned all the Codecs to the SIP to check if the problem was related , but it did not solve the situation.
     
  6. Sopock

    Sopock Member

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    NAT/ALG error check

    what they say on Static Port Mappings?
    but checker will not be able to check private trunk which is similar to my unsupported case?
    Maybe media server is confused and test with single NIC in virtual machine might help?
     
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  7. Trinexus

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    NIC1 is connected to the internet through a Fortigate. It allows the connection with the Voip Provider. There is a route defined on the Fortigate to route all traffic related to the 3CX and the SIP helper service is off.

    The second NIC is used to connect the 3CX server to a different SIP Provider (local Service). It has a static IP (provided by the SIP Provider) and authenticates as a Generic SIP IP Based . It is connected directly to the T-Marc port as per the provider's instructions.

    There is an outgoing rule defined to use the SIP only when a specific number precedes the phone number. It dials out correctly ( I see the DID correctly) and I can hear everything that the recipient says but they can no hear me. When I dial back the SIP DID I get a busy signal.

    The SIP port (in the T-Marc) does not provides internet , so I need both NICs to have the 3cx working - even I have only the SIP and eliminate the Voips.
     

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  8. leejor

    leejor Well-Known Member

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    It may be a case of the audio ports (required) not "listening" on the NIC associated with the trunk that the call goes out on. Simply because you have the outbound rules sending the call out over that trunk, which routes over one NIC, may be no guarantee that the audio ports ports also get associated with that NIC.
     
  9. CentrexJ

    CentrexJ Member

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    Edit the SIP trunk that doesn't work and go to "Options". Make sure the "Select which IP to use in the contact sip and connection sdp fields" is set to the proper IP of the NIC for that trunk.
     
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  10. Trinexus

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    Thank you all. I did placed the IP on the Select which IP to use in 'Contact' (SIP) and 'Connection'(SDP) fields. but did not worked, until I did "loosed" the static route on the server ( PBX address to the gateway on the NIC) by placing 0.0 on the last two entries of the PBX address. It was a little bit tricky but I want to share it just in case someone runs into a situation similar to this.
     
  11. cel

    cel

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    I tried to use within the SIP Trunk/Options the “select which IP to use in contact (SIP) and connection (SDP) fields” a customized, the external IP, but without any success. Anyone an idea. I need to adapt the SDP as my SIP Trunk provider is not via the internet. I go from my NIC to my Firewall and the firewall decides the route for the packets. Outgoing calls are workign fine for me. I have for incoming carrier calls "one way speech" and this is related to the SDP packet which contains the NIC IP instead of the outside IP of the firewall. Any idea?
     
  12. Sopock

    Sopock Member

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  13. Syed123

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    Last year we did 3CX call center installation . Now, we had change premises and to change IP addresses of 3CX server and Patton Gateway. We just changed that part of configuration and we have established outgoing calls, but have problem with incoming calls. That is, for some reason Patton GW is not registering properly to 3CX. Log says its credentials, but I have re-entered and double checked credentials several times on both GW and 3CX (please see attached), but it still says the same thing.
    further i have regenrated the configuration and import it to patton gateway. but the issue is still there.
    Any Help.
     
  14. YiannisH_3CX

    YiannisH_3CX Support Team
    Staff Member 3CX Support

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    Hello @Syed123

    When you local IP changes the only way to make sure that the PBX will work correctly is to un-install and re-install with correct IP.
    If you are on V15 then you need to perform a backup without licence and FQDN, save the backup to a non 3CX location and un-install the system. Then re-install the PBX using the backup and the correct local IP address.
     
  15. Syed123

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    Hi Yiannish

    Thanks for your reply.

    let me do this get back to you.

    Regards
     
  16. Syed123

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    i am sharing another issue here.

    We face a lot of voice breaking issues with 3CX system in UAE (on 3g/4G) or in middle east, secondly if ITSP block SIP ports than in that case what we will do , should we configure TLS ?

    Regards
     
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