VIEWS: 76 PAGES: 8 POSTED ON: 3/10/2010
Make sure you have valid SIP Trunk licences. Create a new SIP trunk. Insert Domain Name, ITSP IP If conference calls and Address, Authentication Name and forwarding calls over SIP is required Password as provided by SIP provider. you must tick RE-INVITE Supported. Tick registration required. Default SIP port, do not change this. Specify over which network topology (LAN port) Specify a codec, Voiceflex supports G711 The SIP trunk will be running over. ALAW and G729. Create a new SIP URI. The local URI field is typically the SIP number that is sent in the invite message from the sip provider. Eg email@example.com is the invite I receive from the provider. 01619251957 is what I would This field is set and cannot be changed. set in the URI field. The URI can be set here or you can set this field to use user data and it will look to the users SIP Tabs for a URI match instead. The Contact and Display Name field is a Create unique incoming and outgoing generic field where you can specify an alternative Group id’s. These will be used to route name and number that will displayed on the receiving Incoming call routes and outgoing SIP calls. handset. Again this field can be set here or on the users SIP tab. Create an IP Route 0.0.0.0 is a default route that will capture Match the Subnet Mask of your all traffic that is not on the IP Office subnet Network. or that doesn’t match another specific IP Route. Define a gateway. This is the local router. Specify over which topology this IP Route is to apply. Create a short code to dial out. The ; in this short code means wait until dialling is complete before sending the invite. The SIP RFC’s state that full number must be present in the invite message for the call to complete. This means you have to either the press # when dialling is complete or set your dial delay count under System/Telephony to 0. This is how to configure the telephone The line group ID relates to the outgoing group number field for a SIP short code. You ID on the SIP URI. use either the domain name or the IP Address of the SIP provider. If you are using the domain name DNS must be tested And working on the IP Office. Create Incoming Call Routes for Incoming calls. The line group ID relates to the incoming Group id on the SIP URI. Set a destination as you normally Would on an incoming call route. Run Stun? If your IP Office is sat behind a NAT device this will break SIP. You may need to run STUN. This will overcome the problems that NAT introduces to SIP. You may want to liaise with your SIP provider to find out if this is a required step or not. Do not fill in any fields in this tab apart from the STUN IP Address, STUN will discover these values for you. Stun server IP address is provided by the SIP provider. All of these values will be discovered by STUN if the process completes successfully. Check this box to re run STUN On start up, ie if there is a power outage etc..
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