Differences

This shows you the differences between two versions of the page.

--- web_gui:sip_switch [2010/11/17 14:36]
tibor
+++ web_gui:sip_switch [2011/09/02 16:03] (current)
tibor
@@ Line 1: / Line 1: @@
+<if ver lt 5.33>
 ====== SIP Switch ======
+(This page is called SIP Switch Advanced in releases 5.33 and later.)
+<else>
+====== SIP Switch Advanced ======
+(This page was called SIP Switch in releases older than 5.33.)
+</if>
 The [[sip:SIP Switch]] gives you PBX-like functions and can give all users of Internet Gate connectivity with the old telephone network, using SIP service provider's single user SIP accounts!
@@ Line 6: / Line 13: @@
 To evaluate the SIP Switch (without having to buy it), press the "Evaluate" link or the "View/Get Example" link below the Dial Plan table! You then get access to the full SIP Switch including 5 user accounts for testing purposes, for a maximum period of 10 days or until the unit is restarted.
+{{ :web_gui:sip_switch.png?267|SIP Switch in rel 5.30}}
 Read more about the [[sip:SIP Switch]].
+[[web GUI:SIP Trunk]]
+[[web GUI:Advanced SIP Settings]]
 ===== Outgoing call number processing =====
@@ Line 58: / Line 71: @@
 ===== SIP Accounts and Incoming Call Processing =====
-The SIP Account table below is used for authentication of SIP users and for extra functionality (columns unlined with pink) that requires the SIP Switch option. Press the "Buy" link below the SIP Account table to buy the SIP Switch or press the "Evaluate" link to make it functional for your testing!
+The SIP Account table is used for authentication of SIP users and for extra functionality (columns unlined with pink) that requires [[:license]] to use.
-Voice mail server - Enter the domain or IP address of the voice mail server here. If more than one domain is entered (separated by space or comma), the next will be tried if the first fails.
+==== Voice mail server ====
+Enter the domain or IP address of the voice mail server. If more than one domain is entered (separated by space or comma), the next will be tried if the first fails.
 If the voice mail server requires specifics in the Request-URI, you can specify it as exemplified:
-o sip:$(cfg.user)@vmserver.com
+  * sip:$(cfg.user)@vmserver.com
-o sip: vmserver.com;mailbox="$(cfg.user)@$(cfg.host)"
+  * sip: vmserver.com;mailbox="$(cfg.user)@$(cfg.host)"
-o sip:$(to.user)@$(to.host);maddr= vmserver.com
+  * sip:$(to.user)@$(to.host);maddr= vmserver.com
 The variables within $() can be:
-- cfg.user = user from SIP Account table
+  * cfg.user = user from SIP Account table
-- cfg.host = host from SIP Account table
+  * cfg.host = host from SIP Account table
-- ruri.user = user from Request-URI
+  * ruri.user = user from Request-URI
-- ruri.host = host from Request-URI
+  * ruri.host = host from Request-URI
-- to.user = user from To header
+  * to.user = user from To header
-- to.host = host from To header
+  * to.host = host from To header
-- from.user = user from From header
+  * from.user = user from From header
-- from.host = host from From header
+  * from.host = host from From header
 and will be replaced with strings from the configuration or SIP message. "user" and "host" refers to the user and host parts in a SIP address sip:user@host.
 Allow external callers to use internal numbers - If checked, an outside caller can use the internal "extension" number to call a local user, e.g. by calling 12@thissipserver.com.
-Account Type - If the Dial Plan forwards using a account as specified in the SIP Account table, it will be processed as specified by the type below. This allows all users of the SIP Switch to e.g. share a single SIP account with connectivity to the ordinary telephone network. All accounts except type "User" can be referenced from dial plan and these accounts are called B2BUA accounts. The full user@host SIP address needs to be specified for B2BUA accounts. Incoming calls can be forwarded to any users as specified in the user action columns.
+==== Accounts Table ====
+  * Account Type - If the Dial Plan forwards using a account as specified in the SIP Account table, it will be processed as specified by the type below. This allows all users of the SIP Switch to e.g. share a single SIP account with connectivity to the ordinary telephone network. All accounts except type "User" can be referenced from dial plan and these accounts are called B2BUA accounts. The full user@host SIP address needs to be specified for B2BUA accounts. Incoming calls can be forwarded to any users as specified in the user action columns.
+    * "User" Ordinary user account with authentication credentials for server side authentication. Not a B2BUA.
+    * "Reg" Register on behalf of client - No SIP phone is connected so the SIP Switch registers the user on the SIP Server.
+    * "XF" Exchange From header - When used by another client, the From header is exchanged to the account owner's.
+    * "XF&Reg" Exchange From header and Register - The From header is exchanged and the SIP Switch registers (both as described above).
+    * "Vng" Vonage - Special type for Vonage SIP accounts with soft PC client. Only phone number (not full URI) needs to be entered as SIP User name.
+    * "Vng&Reg" Vonage and Register - For Vonage SIP accounts without using a Vonage client - the SIP Switch registers.
+    * "MR" Media Relay - In addition to the From header exchange, media is always sent via the SIP Switch.
+    * "MR&Reg" Media Relay and Register - Media is relayed, the From header is exchanged and the SIP Switch registers.
+    * "Domain" Domain with authentication - The SIP Switch authenticates all users e.g. towards a gateway at the specified host address.
+  * Ext. = Internal extension number - Here you specify the internal "extension" number you want to assign to each user. You select any unique number from the reserved range specified above the Dial Plan table.
+  * SIP User Name or SIP Address (URI) - Enter a complete SIP URI or just the username part of a SIP adress. If only the username is set it means that the user belongs to all SIP domains handled by this unit. This is also the User Name you enter when logging into the personal page through "User Log in" at the first web page of this product. (For B2BUA accounts, the full URI always has to be specified). Some attributes can be configured for B2BUA accounts using SIP parameter syntax:
+    * ";hrr" - Hide Record-Route
+    * ";hvia" - Hide Via
+    * ";htt" - Show only one To tag
+    * ";expires=[seconds]" - Sets proposed registration expires value to [seconds]
+    * ";nua" - No User-Agent header added to SIP messages.
+    * ";rhd" - Set domain in (3xx) redirection URI's to home/service domain
+    * ";rcd" - Set domain in (3xx) redirection URI's to caller domain
+    * ";POOL[n]" - assign the B2BUA to pool [n] (see help on SIP Trunk page).
+    * ";max-calls=[nr]" - number of maximum concurrent calls using the B2BUA account.
+  * Authentication User ID - This is the User ID used for authentication of SIP requests. Authentication for registering at this SIP server can be turned on in the General SIP Server Settings above and authentication may also be forced in the Dial Plan. If authentication is turned on, the User ID in the SIP client must match the one entered here. The user can change this field on his personal page.
+  * Authentication Password - This is the Password used for authentication and for logging in to the personal page. If authentication is turned on, the Password in the SIP client must match the one entered here. The user can change this field on his personal page.
+  * Comment - It is useful to enter the user's name here, if he has a numeric SIP name.
+  * Dyn. Regs. - Dynamic Registrations. The number of SIP clients that have been registered for each user is shown here.
+  * Forward Action - Specifies if and how this user's calls are to be forwarded. The user can change this setting on his personal page.
+    * "No" No forwarding is made. Only the Dynamically Registered clients will ring.
+    * "Forward" The Dynamically Registered client(s) will not ring - only the one(s) listed in the next column will ring.
+    * "Parallel" All SIP clients will ring simultaneously, both the Dynamically Registered and the ones specified in the next column.
+    * "Sequence" The SIP clients will ring one after another, for 25 seconds each until there is an answer. The Dynamically Registered will ring first, followed by the ones specified in the next column in the given order. You can control the ringing order by adding the q-parameter after the SIP user name. q is a value between 0 and 1 and if no value is given, q=0.5 is assumed. SIP clients with the highest q-value, e.g. "john@company.com;q=0.6", will ring first. The default ring length will be overridden if another time is specified in the Voice Mail column. You can also set the ring length in seconds by adding the t-parameter after a user name, e.g. "john@company.com;t=15" will make john ring for 15 seconds.
+    * "Random" All SIP clients will ring in random order, both the Dynamically Registered and the ones specified in the next column.
+    * "Deny" means that this user will not accept any calls at all.
+  * to SIP URI, IP Address, Phone Number or Ext. - Here you specify the SIP PC clients, the SIP phones or the PSTN phones that the calls for this specific user will be forwarded to. An entered phone number will be processed by the Dial Plan. For a SIP phone on the same LAN using an external SIP server, it is recommended that you specify the internal extension number or the local IP address (e.g. john@192.168.0.36 instead of john@company.com) to avoid restrictions in the external SIP server. You can enter several receivers separated by space or comma. The user can change this field on his personal page.
+  * Voice Mail - Here you select whether, and on which condition, this user's calls shall be redirected to the Voice mail server (specified above this table).
+  * BU = Block User - When checked, this user will not be allowed to log in to his personal page to change his settings. The personal page is reached through the "User Log in" on the first web page of this product. You use the SIP name of column 3 and the Password of column 5 to log in.
+  * RI = Restrict Incoming - By checking this box, the user will only be reachable for callers defined by the "Allowed Incoming Callers" lists. Also see next point!
+  * AC = Accept Common - There is both a personal list (controlled from the personal page, see BU above) and a common list (on this page) to specify the Allowed Incoming Callers. If RI is selected, then this box selects whether the common list on this page will be used in addition to the user's own list.
+  * Incoming Call Matching - A PSTN gateway may pass an incoming call with the dialled number in the Request-URI. Here you can match one or several incoming PSTN numbers (separated by space or comma) to specific users. If the account is of "Reg"-type, the SIP Switch will use the number entered here in the contact header when registering.
+  * Show other dynamically registered SIP users - There may be other SIP users registered, that you have not created a SIP account for (which is not necessary). You can view these by checking this box. When viewing these, you can mark and add such users to the SIP Account table.
+===== Incoming Call Blacklist =====
+Users with SIP addresses listed here are not allowed make incoming calls or otherwise use this SIP server. The SIP addresses may include wildcards. ? represents any single character while * represents a string of characters of any length. * is only allowed first, last and just before or after @ (e.g. sex*@* and *@sex*). Each user can also set up his own blacklist in addition to this one, on his personal page.
+:!: It is easy for a caller to change SIP address, so this method is quite easy to bypass.
+===== Allowed Incoming Callers =====
+When the RI box in the SIP Account table is checked, no calls are allowed for that user unless specified in the personal "Allow Incoming Callers" section or, when AC is checked, unless specified in the common "Allow Incoming Callers" section here.
+  * Local users (on this server) - Here you can select if calls from users handled by this SIP server should be allowed or allowed only after authentication. :!: Note that it is not difficult for a non-authorized user to assume a local user's SIP address (spoof) and make incoming calls when Authentication is not selected.
+  * Authenticated - Callers with any of the entered User IDs and Passwords are allowed.
+  * Not Authenticated - Callers with any of the SIP addresses entered here are allowed. You can use wildcards to specify allowed callers. ? represents any single character while * represents a string of characters of any length. * is only allowed first, last and just before or after @ (e.g. *@partner.com).
+===== Export/Import Settings =====
+You can save the settings on this page as a file on you hard disk by pressing Export. When Importing settings from a previously stored file, you can select which parts of the settings you want to restore. The specific SIP Switch Settings will not be restored if you have not purchased the SIP Switch.
+:!: It is recommended that you make a security back-up of your settings by Exporting them to a file on your hard disk!
-"User" Ordinary user account with authentication credentials for server side authentication. Not a B2BUA.
+======  ======
-"Reg" Register on behalf of client - No SIP phone is connected so the SIP Switch registers the user on the SIP Server.
-"XF" Exchange From header - When used by another client, the From header is exchanged to the account owner's.
-"XF&Reg" Exchange From header and Register - The From header is exchanged and the SIP Switch registers (both as described above).
-"Vng" Vonage - Special type for Vonage SIP accounts with soft PC client. Only phone number (not full URI) needs to be entered as SIP User name.
-"Vng&Reg" Vonage and Register - For Vonage SIP accounts without using a Vonage client - the SIP Switch registers.
-"MR" Media Relay - In addition to the From header exchange, media is always sent via the SIP Switch.
-"MR&Reg" Media Relay and Register - Media is relayed, the From header is exchanged and the SIP Switch registers.
-"Domain" Domain with authentication - The SIP Switch authenticates all users e.g. towards a gateway at the specified host address.
-Ext. = Internal extension number - Here you specify the internal "extension" number you want to assign to each user. You select any unique number from the reserved range specified above the Dial Plan table.
-SIP User Name or SIP Address (URI) - Enter a complete SIP URI or just the username part of a SIP adress. If only the username is set it means that the user belongs to all SIP domains handled by this unit. This is also the User Name you enter when logging into the personal page through "User Log in" at the first web page of this product. (For B2BUA accounts, the full URI always has to be specified). Some attributes can be configured for B2BUA accounts using SIP parameter syntax:
-- ";hrr" - Hide Record-Route
-- ";hvia" - Hide Via
-- ";htt" - Show only one To tag
-- ";expires=[seconds]" - Sets proposed registration expires value to [seconds]
-- ";nua" - No User-Agent header added to SIP messages.
-- ";rhd" - Set domain in (3xx) redirection URI's to home/service domain
-- ";rcd" - Set domain in (3xx) redirection URI's to caller domain
-- ";POOL[n]" - assign the B2BUA to pool [n] (see help on SIP Trunk page).
-- ";max-calls=[nr]" - number of maximum concurrent calls using the B2BUA account.
-Authentication User ID - This is the User ID used for authentication of SIP requests. Authentication for registering at this SIP server can be turned on in the General SIP Server Settings above and authentication may also be forced in the Dial Plan. If authentication is turned on, the User ID in the SIP client must match the one entered here. The user can change this field on his personal page.
-Authentication Password - This is the Password used for authentication and for logging in to the personal page. If authentication is turned on, the Password in the SIP client must match the one entered here. The user can change this field on his personal page.
-Comment - It is useful to enter the user's name here, if he has a numeric SIP name.
-Dyn. Regs. - Dynamic Registrations. The number of SIP clients that have been registered for each user is shown here.
-Forward Action - Specifies if and how this user's calls are to be forwarded. The user can change this setting on his personal page.
+[[web GUI:SIP Trunk]]
-"No" No forwarding is made. Only the Dynamically Registered clients will ring.
-"Forward" The Dynamically Registered client(s) will not ring - only the one(s) listed in the next column will ring.
-"Parallel" All SIP clients will ring simultaneously, both the Dynamically Registered and the ones specified in the next column.
-"Sequence" The SIP clients will ring one after another, for 25 seconds each until there is an answer. The Dynamically Registered will ring first, followed by the ones specified in the next column in the given order. You can control the ringing order by adding the q-parameter after the SIP user name. q is a value between 0 and 1 and if no value is given, q=0.5 is assumed. SIP clients with the highest q-value, e.g. "john@company.com;q=0.6", will ring first. The default ring length will be overridden if another time is specified in the Voice Mail column. You can also set the ring length in seconds by adding the t-parameter after a user name, e.g. "john@company.com;t=15" will make john ring for 15 seconds.
-"Random" All SIP clients will ring in random order, both the Dynamically Registered and the ones specified in the next column.
-"Deny" means that this user will not accept any calls at all.
-to SIP URI, IP Address, Phone Number or Ext. - Here you specify the SIP PC clients, the SIP phones or the PSTN phones that the calls for this specific user will be forwarded to. An entered phone number will be processed by the Dial Plan. For a SIP phone on the same LAN using an external SIP server, it is recommended that you specify the internal extension number or the local IP address (e.g. john@192.168.0.36 instead of john@company.com) to avoid restrictions in the external SIP server. You can enter several receivers separated by space or comma. The user can change this field on his personal page.
-Voice Mail - Here you select whether, and on which condition, this user's calls shall be redirected to the Voice mail server (specified above this table).
-BU = Block User - When checked, this user will not be allowed to log in to his personal page to change his settings. The personal page is reached through the "User Log in" on the first web page of this product. You use the SIP name of column 3 and the Password of column 5 to log in.
+[[web GUI:Advanced SIP Settings]]
-RI = Restrict Incoming - By checking this box, the user will only be reachable for callers defined by the "Allowed Incoming Callers" lists. Also see next point!
-AC = Accept Common - There is both a personal list (controlled from the personal page, see BU above) and a common list (on this page) to specify the Allowed Incoming Callers. If RI is selected, then this box selects whether the common list on this page will be used in addition to the user's own list.
-Incoming Call Matching - A PSTN gateway may pass an incoming call with the dialled number in the Request-URI. Here you can match one or several incoming PSTN numbers (separated by space or comma) to specific users. If the account is of "Reg"-type, the SIP Switch will use the number entered here in the contact header when registering.
-Show other dynamically registered SIP users - There may be other SIP users registered, that you have not created a SIP account for (which is not necessary). You can view these by checking this box. When viewing these, you can mark and add such users to the SIP Account table.

web_gui/sip_switch.1290000995.txt.gz · Last modified: 2010/11/17 14:36 by tibor

Show page Old revisions