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Technical FAQs

Here are some of the frequently asked questions regarding the technical aspects of setting up and using your SIP Trunking account. Click the links below to learn more. If you need further assistance, click here to contact us.

  1. Does this service work with any type of IP-PBX?
  2. What SIP Proxy should the gateway point to?
  3. Can a different port other than UDP port 5060 be used?
  4. What is the registration string for the gateway?
  5. Can a dynamic IP address connect to Net2Phone services?
  6. Everything is set up correctly but cannot pass traffic. How can this be fixed?
  7. How can one way or no audio issues be fixed?
  8. What Codecs are supported?
  9. Is there a recommended Codec to use with Net2Phone service?
  10. Does Net2Phone pass CLI?
  11. What format should be used when setting up Caller ID in the switch?
  12. Can a numberthat is not provided by Net2Phone be sent in the caller ID?
  13. Which dialing plan should be used when placing calls?
  14. Does this service support Fax transmissions?
  15. Does Net2Phone support VPN connectivity?
  16. Does Net2Phone support SIP over TLS?
  17. Does the switch need to support SIP re-INVITEs?
  18. How can a SIP trace be captured on the switch?
  19. Does Net2Phone support SRTP?
  20. Does Net2Phone support point-to-point connectivity from the network to the Net2Phone data center?
  21. Are there any other web sites which offer help in setting up the PBX?

  1. Does this service work with any type of IP-PBX?
    2. Net2Phone’s Business VoIP service works with any IP enabled PBX. Set up instructions might be slightly different by brand and operating system but as long as the PBX can support passing IP traffic, Net2Phone can pass your traffic.

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  2. What SIP Proxy should the gateway point to?
    3. Set the SIP proxy for this service to point toippbx.net2phone.com UDP port 5060. Below is a sample configuration for a typical Asterisk gateway. The same parameters will apply for other types of gateways but the configuration may look slightly different. Please consult with the gateway vendor or software provider if unsure how to configure the below parameters.
    On open source applications such as Asterisk, setup your SIP trunk as follows.
    Please note that no SIP registration is required against Net2Phone's platform.

    General
    port=5060
    dtmfmode=rfc2833
    Progressinband=never
    allow=g729
    allow=g711
    allow=g723

    Outbound Calling
    type=peer
    username=ACCOUNT NUMBER
    secret=PIN NUMBER
    host=ippbx.net2phone.com
    port=5060
    canreinvite=no

    Registration
    register=>ACCOUNT NUMBER:PIN NUMBER@ippbx.net2phone.com/siptrunking

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  3. Can a different port other than UDP port 5060 be used?
    4. UDP port 5061 or UDP port 6070 can be used as well.

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  4. What is the registration string for the gateway?
    5. register=>ACCOUNT NUMBER:PINNUMBER@ippbx.net2phone.com/siptrunking

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  5. Can a dynamic IP address connect to Net2Phone services?
    6. Yes. Net2Phone uses account and pin authentication rather than IP addresses therefore we do not need to know the IP Address.

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  6. Everything is set up correctly but cannot pass traffic. How can this be fixed?
    7. First check that the gateway is being pointed at the correct SIP proxy and the correct port (see above). If that is correct then check for any denies in the firewall logs for SIP traffic. It is possible that there might be unintentional blocking of UDP 5060 traffic in the firewall to / from the SIP proxy sever or there might be an access-list on the edge router that is not allowing the SIP traffic to pass to / from the gateway. If the IP-PBX Gateway is behind a NAT and the gateway is not registering with Net2Phone, check the NAT rules in the router to make sure that the SIP traffic is reaching the private network from the public network. If possible, check the SIP logs on the gateway to see if you are getting any SIP replies from Net2Phone.

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  7. How can one way or no audio issues be fixed?
    8. Check to make sure RTP (media) traffic is not blocked to our proxy servers. Generally speaking, it is recommended that RTP traffic is open to the internet and to not filter media traffic in any way.

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  8. What Codecs are supported?
    9. G711u-law, G729, G723 codecs are supported.

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  9. Is there a recommended Codec to use with Net2Phone service?
    10. It is recommended to offer multiple codecs in the SIP INVITE to Net2Phone (G729, G711, G723). If all codecs can not be offered, then it is preferred if to offer G729 only.

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  10. Does Net2Phone pass CLI?
    11. Net2Phone makes the best effort to pass CLI but CLI is not guaranteed on this service.

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  11. What format should be used when setting up Caller ID in the switch?
    12. Net2Phone customers are preferred to use the PAID (P-Asserted-Identity) option for CLI as per RFC 3325. The Net2Phone platform also supports RPID (Remote-Party-ID) as well. http://www.ietf.org/rfc/rfc3325.txt

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  12. Can a numberthat is not provided by Net2Phone be sent in the caller ID?
    13. Yes.

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  13. Which dialing plan should be used when placing calls?
    14. NANP format ( 1+10 digit on US calls and 011 + CC+ on international calls) should be used. For more information on North American dial plan formatting, visit www.nanpa.com & for more information regarding E164 formatting, view the following PDF online at www.itu.int/dms_pub/itu-t/opb/sp/T-SP-E.164D-2009-PDF-E.pdf.

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  14. Does this service support Fax transmissions?
    15. Fax is supported but not guaranteed. Please make sure that G711 codec is being used when sending faxes.

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  15. Does Net2Phone support VPN connectivity?
    16. Currently, there are no VPN solutions for the Business VoIP service.

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  16. Does Net2Phone support SIP over TLS?
    17. SIP over Transport Layer Security is not currently supported.

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  17. Does the switch need to support SIP re-INVITEs?
    18. There is a 15 minute SIP session timer set on all incoming traffic which means that Net2Phone will send a SIP re-INVITE to the switch 15 minutes into the call. Please make sure that the switch is allowing SIP re-INVITEs. If unsure how to enable SIP re-INIVEs on a specific switch, please reach out to the vendor.

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  18. How can a SIP trace be captured on the switch?
    19. When setting up a new SIP trunk with a provider or troubleshooting call failures it is important to be able to capture a signaling trace of an outbound call. This is also important when troubleshooting SIP registration issues with a new provider.

    In Asterisk for example, turn on SIP debugging via the Asterisk CLI using the sip set debug commands.

    Sip set debug peer on - turns on SIP debugging globally showing all SIP traffic to and from the Asterisk gateway.

    Sip set debug IP xxx.xxx.xxx.xxx - debug only message to and from a particular IP address.

    Sip set debug off – Turns off all SIP debugging.

    Please be familiar with how to turn on debugging.

    If enabling debugging on the switch is not preferred, use a network protocol analyzer such as Wireshark to capture the SIP and media traffic on the calls. To learn more about Wireshark, please visit http://www.wireshark.org/ This site includes step by step videos on how to setup Wireshark on a network.

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  19. Does Net2Phone support SRTP?
    20. Secure Real Time Transport Protocol is not currently supported.

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  20. Does Net2Phone support point-to-point connectivity from the network to the Net2Phone data center?
    21. To establish point to point connectivity to the Net2phone platform instead of routing the traffic over the public internet, please inquire with a Net2phone Account Manager via reseller@net2phone.com.

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  21. Are there any other web sites which offer help in setting up the PBX?
    22. It is important to note that Net2Phone does not take any responsibility for what these web sites have to offer or if the information provided on these sites is accurate, but the following sites tend to be helpful.
    Asterisk Support Forum - http://forums.asterisk.org/
    3CX Support Forum - http://www.3cx.com/forums/
    Avaya Support Forum - http://support.avaya.com/css/appmanager/public/support
    Elastix Support Forum - http://www.elastix.org/index.php/en/component/kunena/
    Trixbox Support Forum - http://fonality.com/trixbox/forum
    Grandstream Support Forum - http://forums.grandstream.com/forums/
    Cisco Support Forum - http://www.cisco.com/cisco/web/support/index.html
    FreePBX Support Forum - http://www.freepbx.org/forums
    Talkswitch Support Forum - http://www.talkswitch.com/us/en/support/
    AudioCodes Support - http://www.audiocodes.com/support
    Voipswitch Support - http://www.voipswitch.com/
    IPsmarx Support - http://www.ipsmarx.com/Support.html
    Huawei  Support - http://support.huawei.com/support/
    Sonus Networks Support - http://sonusnetworks.force.com/PortalLoginPage
    PortaOne Support - http://portaone.com/support/portacare/

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