Knowledge Base

• For questions regarding per minute rates, phone number availability, or signing up for an account, please contact us. • For questions regarding your account, contact us or contact chat support. • For any other inquiries, you can contact a sales representative by starting a chat or filling out the contact form.

Secure Real Time Transport Protocol is not currently supported.

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.

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.

SIP over Transport Layer Security is not currently supported.

Currently, there are no VPN solutions for the Business VoIP service.

Fax is supported but not guaranteed. Please make sure that G711 codec is being used when sending faxes.

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.

Yes

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

It is recommended to offer multiple codecs in the SIP INVITE to Switch2Voip (G729, G711, G723). If all codecs can not be offered, then it is preferred if to offer G729 only.

G711u-law, G729, G723 codecs are supported.

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.

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 server 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.

Yes. Switch2Voip uses account and pin authentication rather than IP addresses therefore we do not need to know the IP Address.

register=>ACCOUNT NUMBER:PIN NUMBER@sip.switch2voip.us/siptrunking

UDP port 5061 or UDP port 6070 can be used as well.

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 Outbound Calling type=peer username=USERNAME secret=PASSWORD host=sip.switch2voip.us port=5060 canreinvite=no Registration register=>USERNAME:PASSWORD@sip.switch2voip.us/siptrunking

Yes, Switch2Voip offers the option of as many phone numbers as preferred on the account. Please contact our support to add phone numbers.

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.