This section will show how to use a SIP trunk to connect CME to a VoIP service provider over the Internet. In this example, long distance dialing will be through the VoIP service provider and e911 services.
The normal protocol that CME IP phones use is SCCP (Skinny Call Control Protocol) but to connect to service providers, the Session Initiation Protocol (SIP) is typically used.
Enabling SIP connections
I've created SIP trunks without enabling SIP to SIP communications but it is better to add the following lines. Sometimes or I should often, there is an incompatibility between implementations of SIP and you may have to disable 302 sending and REFER sending.
voice service voip allow-connections sip to sip no supplementary-service sip moved-temporarily !---Disable 302 sending no supplementary-service sip refer !---Disable REFER sending
2. Authenticating to the Service Provider
A SIP user agent is configured with the username and password that has been provided by the VoIP service provider. In this example, the username is "MyUsername" and the password is "MyPassword". A fictious domain "voip.isp.ex" is used for this example for the VoIP service provider domain name.
sip-ua authentication username MyUsername password MyPassword no remote-party-id retry invite 2 retry register 10 retry options 0 timers connect 100 registrar dns:voip.isp.ex expires 3600 sip-server dns:voip.isp.ex host-registrar
3. Create a Long Distance Dialplan
A dial-peer is created to use VoIP and the SIP protocol. The destination pattern for long distance dialing is dial 1 then the following 10 digits are sent to the VoIP service provider. The target is the fictious voip.isp.ex service provider. You can specify the codec to use, in this case it is g711ulaw (64kbps) but if you have bandwidth restrictions, you can select others: g.729 (8kpbs but requires a license), iLBC (Internet low bit rate 11 kbps) or a higher quality codec g.722 (64kbps but better quality audio). Note, the service provider must agree on the codec to use!
dial-peer voice 100 voip destination-pattern 1.......... session protocol sipv2 session target dns:voip.isp.ex dtmf-relay rtp-nte codec g711ulaw no vad
"dtmf-relay rtp-nte" specifies the method used to transmit the dual tone multi frequency (dtmf) tones used when pushing the IP phone's keypad buttons.
Dialing 1 + a 10 digit long distance number should dial out through the VoIP service provider.
4. Create an e911 Dialplan
This is a simple e911 emergency dialplan that directly dials the VoIP service provider's e911 services. Call Manager Express provides Enhanced 911 services with Emergency Response Zones for large multibuilding/multilocation installations. The Enhanced 911 services is beyond the scope of this webpage.
A dial-plan is created so that when 911 is dialed, it connects directly to the VoIP service provider's e911 services. The VoIP service provider must have up to date information on the address of the caller. There has been several tragic cases where the emergency services have responded to an old address during an emergency call and a life was lost due to the delay in sorting out where the emergency call actually originated from! Make sure your address information is up to date!
dial-peer voice 911 voip destination-pattern 911 session protocol sipv2 session target dns:voip.isp.ex dtmf-relay rtp-nte codec g711ulaw no vad
Dialing 911 should dial out to the VoIP service provider and the service provider should recognize the 911 DID and your CID and direct the call to emergency services for your area.
If this page has helped you, please consider donating $1.00 to support the cost of hosting this site, thanks.