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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.17s
  • Memory usage: 2.18MB
  • Database queries: 32
  • GZIP: Disabled
  • Server load: 2.89

SER module group

group: This module export functions necesarry for group membership checking. There is a database table that contains list of users and groups they belong to. The table is used by functions of this module.

A typical use is to check whether the user is the member of a particular group (such as LongDistance), in order to control access to a particular facility.

The is_user_in(URI, group) function allows checking whether the user is a member of a specifiic group. The 'URI' parameter specifies the field which contains the URI. It can be one of the following values:
  • Request-URI
  • To
  • From
  • Credentials

Configuration


The Group module must be loaded after a database module (such as mysql). It nneds to be told the tables and columns to use when checking group membership.

  • "db_url" specifies the database connection to use. Default 'sql://serro:47serro11@localhost/ser"'
  modparam("group", "db_url", "sql://username:password@dbhost/ser")
  • "table" specifies the database table to use. Default 'grp'
  modparam("group", "table", "grp")
  • "user_column" specifies the column which conatins the username. Default 'username'
  modparam("group", "user_column", "username")
  • "domain_column" specifies the column which contains the SIP domain. Default 'domain'
  modparam("group", "domain_column", "domain")
  • "group_column" specifies the column which contains the group name. Default 'grp'
  modparam("group", "group_column", "grp")
  • "use_domain" is an integer. '1' specifies match "user@domain", while '0' specifies match 'user'
  modparam("group", "use_domain", 1)

Example

Check whether a user is allowed access to long-distance calls. Since the user URI is being extracted from "credentials" it means that they have already been authenticated.

  if (!is_user_in("credentials", "ld")) { 
     sl_send_reply("403", "Local calls only");
     break;
  };



Back to SIP Express Router
Created by oej, Last modification by papafox on Thu 19 of Aug, 2004 [07:02 UTC]

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