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Differences
This shows you the differences between two versions of the page.
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web_gui:sip_trunk [2010/11/18 16:00] tibor |
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| ^ :!: You might need to purchase a [[:license]] to be able to use SIP Trunks in your unit. ^ | ^ :!: You might need to purchase a [[:license]] to be able to use SIP Trunks in your unit. ^ | ||
| | Without SIP Trunk license the SIP Trunk page is not functional. | | | Without SIP Trunk license the SIP Trunk page is not functional. | | ||
| + | |||
| + | Read more about [[SIP:Trunk|SIP Trunks]]. | ||
| + | |||
| + | {{ :web_gui:sip_trunk.png?231|SIP Trunk in rel 5.30}} | ||
| ===== Select SIP Trunk ===== | ===== Select SIP Trunk ===== | ||
| Line 61: | Line 65: | ||
| * Forward to - Optional, for PBX Lines this the number incoming calls will be forwarded to on the PBX. For SIP Lines this is the complete SIP URI an incoming call will be forwarded to or a SIP user name/number on this unit's SIP Server. Leave this fiels empty to enable a DID function (several numbers on the same trunk line). If it is empty, the number/user of the incoming call will not be changed when forwarding the call. | * Forward to - Optional, for PBX Lines this the number incoming calls will be forwarded to on the PBX. For SIP Lines this is the complete SIP URI an incoming call will be forwarded to or a SIP user name/number on this unit's SIP Server. Leave this fiels empty to enable a DID function (several numbers on the same trunk line). If it is empty, the number/user of the incoming call will not be changed when forwarding the call. | ||
| * Ext. - Optional, if this unit has the SIP Server license installed, this is an extension number that can be used by other SIP devices on this unit to reach the PBX on this line. Only for PBX Lines and Main Trunk Line as SIP users have their extension number configured on the SIP Switch page. | * Ext. - Optional, if this unit has the SIP Server license installed, this is an extension number that can be used by other SIP devices on this unit to reach the PBX on this line. Only for PBX Lines and Main Trunk Line as SIP users have their extension number configured on the SIP Switch page. | ||
| + | |||
| + | ===== Setup for the PBX ===== | ||
| + | This section deals with settings for the PBX, where it is located and how it is contacted on SIP. | ||
| + | * Use settings - Choose whether the settings for PBX should be taken from this page or another SIP Trunk page. | ||
| + | * PBX Name - A human readable comment that lets you put a name on this PBX. | ||
| + | * Use alias IP address - Optional, if IP aliases have been configured on Network Configuration page you can select one of them for usage when communicating with the PBX. | ||
| + | * PBX SIP Address - Optional, if the PBX register on this unit, this is the SIP account (address-of-record) it registers to. | ||
| + | * User ID - Optional, the user ID for digest authentication of the PBX. | ||
| + | * Password - Optional, the password for digest authentication of the PBX. | ||
| + | * PBX IP address - Optional, the IP address of the PBX. Required if the PBX does not register. | ||
| + | * PBX Domain Name - Optional, the SIP domain name of the PBX in case the PBX wants incoming calls be addressed to %%sip:number@domain%% instead of %%sip:number@ip-address%% | ||
| + | * Forward incoming calls to PBX using - Choose how the PBX is located: | ||
| + | * IP Address = incoming calls are forwarded to "PBX IP Address". | ||
| + | * Domain name = incoming calls are forwarded to "PBX Domain Name". | ||
| + | * Registered Address = incoming calls are forwarded to the registered binding of the PBX. | ||
| + | * Signaling transport - The transport protocol for SIP messages sent to the PBX. Automatic means the transport protocol is determined automatically by applying the rules of [[http://www.ietf.org/rfc/rfc3263.txt|RFC 3263]] on the SIP URI sent to PBX. | ||
| + | * Port number - Optional, the destination port number to use for SIP requests sent to the PBX. If not specified the port number is determined automatically by applying the rules in [[http://www.ietf.org/rfc/rfc3263.txt|RFC 3263]] on the SIP URI sent to the PBX. | ||
| + | * SIP host name(s) used by PBX - Optional, the host or domain name used in the From SIP URI by the PBX for outgoing calls. Required if the PBX does not put a local IP address in the From Field. | ||
| + | * Match From Number/User in field - Sets which field in the SIP message that will used as the callers number when matching the From * Number/User column in the line specifications above. | ||
| + | |||
| + | * To header field - Set the To header field to use for incoming calls to the PBX: | ||
| + | * Same as Request-URI = The To header field is set to the same value as the Request-URI. | ||
| + | * Copy from Trunk = Copy the To URI from the incoming call on the Trunk interface. | ||
| + | * Initial Request-URI = Set To equaling the Request-URI as it looks initially before passing internal SIP proxy. | ||
| + | * As entered = Enter the URI to use in To header manually in the box to the right. Variable substituion as decribed above is available in this box. | ||
| + | * Trunk Group Parameters usage - Optional, this setting makes it possible to use the Trunk Group Parameters defined above also on the PBX interface. | ||
| + | * Originating Trunk Group Parameters = The TGP will be used for signalling originating TGP when forwarding calls to the PBX. | ||
| + | * Destination Trunk Group Parameters = The TGP will be used for matching destination TGP on outgoing calls. If they match the configuration in this page will be used regardless of what is written is Dial Plan of SIP Switch page. If there is no match, nothing happens. | ||
| + | * Originating and Destination T.G.P = The TGP will be used for both of the above. | ||
| + | |||
