Switch2Voip’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, Switch2Voip can pass your traffic.
Set the SIP proxy for this service should point to sip.switch2voip.us 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 Switch2Voip's platform.
General port=5060 dtmfmode=rfc2833 Progressinband=never allow=g729 allow=g711 allow=g723
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 Switch2Voip, 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 Switch2Voip.
Switch2Voip customers are preferred to use the PAID (P-Asserted-Identity) option for CLI as per RFC 3325. The Switch2Voip platform also supports RPID (Remote-Party-ID) as well. http://www.ietf.org/rfc/rfc3325.txt
There is a 15 minute SIP session timer set on all incoming traffic which means that Switch2Voip 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.
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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Yes. Choose a phone number in the U.S., Canada, or over 40 other countries. For example, if a U.S. phone number is selected and the user lives outside the U.S., a caller will have the same experience as calling any other U.S. phone number!