pua dialoginfo

Anca-Maria Vamanu

Edited by

Anca-Maria Vamanu

Klaus Darilion

Edited by

Klaus Darilion

   Copyright © 2006 Voice Sistem SRL

   Copyright © 2008 Klaus Darilion IPCom
   Revision History
   Revision $Revision$ $Date: 2013-01-29 14:35:11 +0200
                              (Tue, 29 Jan 2013) $
     __________________________________________________________

   Table of Contents

   1. Admin Guide

        1.1. Overview
        1.2. Dependencies

              1.2.1. OpenSIPS Modules
              1.2.2. External Libraries or Applications

        1.3. Exported Parameters

              1.3.1. include_callid (int)
              1.3.2. include_tags (int)
              1.3.3. include_localremote (int)
              1.3.4. caller_confirmed (int)
              1.3.5. presence_server (string)
              1.3.6. caller_spec_param (string)
              1.3.7. callee_spec_param (string)
              1.3.8. osips_ps (int)

        1.4. Exported Functions

              1.4.1. dialoginfo_set([side])

   List of Examples

   1.1. Set include_callid parameter
   1.2. Set include_tags parameter
   1.3. Set include_localremote parameter
   1.4. Set caller_confirmed parameter
   1.5. Set presence_server parameter
   1.6. Set caller_spec_param parameter
   1.7. Set caller_spec_param parameter
   1.8. Set osips_ps parameter
   1.9. dialoginfo_set usage

Chapter 1. Admin Guide

1.1. Overview

   The pua_dialoginfo retrieves dialog state information from the
   dialog module and PUBLISHes the dialog-information using the
   pua module. Thus, in combination with the presence_xml module
   this can be used to derive dialog-info from the dialog module
   and NOTIFY the subscribed watchers about dialog-info changes.
   This can be used for example with SNOM and Linksys phones.

   Note: This implements dialog-info according to RFC 4235 and is
   not compatible with the BLA feature defined in
   draft-anil-sipping-bla-03.txt. (Actually the BLA draft is
   really crap as it changes SIP semantics)

   The module is based on code (copy/paste) from pua_usrloc and
   nat_traversal module.

   Following you will show some examples of an dialog-info XML
   document taken from RFC 4235. This will help you to understand
   the meaning of the module parameters:

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info"
             version="1"
             state="full"
             entity="sip:alice@example.com">
    
        <state>early</state>
    
</dialog-info>

   The root element is the "dialog-info". It contains the
   namespace, the version (which must be incremented for each new
   PUBLISH for this certain dialog), the state (this module only
   supports state=full) and the entity for which we publish the
   dialog-info.

   The "dialog" element must contain an id parameter. The id
   parameter is usually different to the optional call-id
   parameter (which is the call-id of the INVITE request) as an
   INVITE can create multiple dialogs (forked request). But as the
   dialog module does not support multiple dialogs created by a
   single transaction, the pua_dialoginfo module sets the id
   parameter to the same value as the call-id parameter. The
   "local-tag" indicates the local tag of the entity. The
   remote-tag indicates the tag of the remote party. The
   "direction" indicates if the entity was the initator of the
   dialog or the recepient (aka if the entity sent or received the
   first INVITE).

   The "state" element describes the state of the dialog state
   machine and must be either: trying, proceeding, early,
   confirmed or terminated.

   The dialog element can contain optional "local" and "remote"
   elements which describes the local and the remote party in more
   detail, for example:

<?xml version="1.0" encoding="UTF-8"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info"
             version="1" state="full">
    
        <state>early</state>
        <local>
            <identity display="Alice">sip:alice@example.com</identity>
            <target uri="sip:alice@phone11.example.com"/>
        </local>
        <remote>
            <identity display="Bob">sip:bob@example.org</identity>
            <target uri="sip:bobster@phone21.example.org"/>
        </remote>
    
</dialog-info>

   The local and remote elements are needed to implement call
   pickup. For example if the above XML document is received by
   somebody who SUBSCRIBEd the dialog-info of Alice, then it can
   pick-up the call by sending an INVITE to Bob (actually I am not
   sure if it should use the URI in the identity element or the
   URI in the target parameter) which contains a Replaces header
   which contains the call-id and the tags. This was tested
   sucessfully with Linksys SPA962 phones and with SNOM 320
   Firmware 7.3.7 (you have to set the function key to
   "Extension").

   A dialog-info XML document may contain multiple "dialog"
   elements, for example if the entity has multiple ongoing
   dialogs. For example the following XML document shows a
   confirmed dialog and an early (probably a second incoming call)
   dialog.

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info"
             version="3"
             state="full"
             entity="sip:alice@example.com">
    
        <state>confirmed</state>

        <state>early</state>
    
</dialog-info>

   To enable dialoginfo notifications for a certain dialog, you
   must call dialoginfo_set() function for that dialog. This
   function can take one parameter which through which you can
   tell the module to publish dialoginfo only for one side of the
   call. This is useful because you want to store dialoginfo only
   for the local users, and you can decide from the script if the
   call parties are local users and give the correct parameter to
   this function to tell it to send generate dialoginfo only for
   the local users. The possible values are : "A" - corresponding
   to generate dialoginfo only for the caller and "B" - generate
   dialoginfo only for the callee. If no parameter is given, the
   module will generate dialoginfo for both parties. It is
   possible to specify what URIs should be used for caller and
   callee by setting the the pseudovariables with the names
   defined as module parameter "caller_spec_param" and
   "callee_spec_param" before calling dialoginfo_set function.
   Please read the description of this parameters in Exported
   Parameters section. If this parameters are not set, the default
   sources will be used, From header for the caller and display
   name in To header + RURI for the callee.

   As the dialog module callbacks only address a certain dialog,
   the pua_dialoginfo always PUBLISHes XML documents with a single
   "dialog" element. If an entity has multiple concurrent dialogs,
   the pua_dialoginfo module will send PUBLISH for each dialog.
   These multiple "presenties" can be aggregated by the
   presence_dialoginfo module into a single XML document with
   multiple "dialog" elements. Please see the description of the
   presence_dialoginfo module for details about the aggregation.

   If there are problems with the callbacks from dialog module and
   you want to debug them you define PUA_DIALOGINFO_DEBUG in
   pua_dialoginfo.c and recompile.

1.2. Dependencies

1.2.1. OpenSIPS Modules

   The following modules must be loaded before this module:
     * dialog.
     * pua.

1.2.2. External Libraries or Applications

   The following libraries or applications must be installed
   before running OpenSIPS with this module loaded:
     * libxml.

1.3. Exported Parameters

1.3.1. include_callid (int)

   If this parameter is set, the optional call-id will be put into
   the dialog element. This is needed for call-pickup features.

   Default value is “1”.

   Example 1.1. Set include_callid parameter
...
modparam("pua_dialoginfo", "include_callid", 0)
...

1.3.2. include_tags (int)

   If this parameter is set, the local and remote tag will be put
   into the dialog element. This is needed for call-pickup
   features.

   Default value is “1”.

   Example 1.2. Set include_tags parameter
...
modparam("pua_dialoginfo", "include_tags", 0)
...

1.3.3. include_localremote (int)

   If this parameter is set, the optional local and remote
   elements will be put into the dialog element. This is needed
   for call-pickup features.

   Default value is “1”.

   Example 1.3. Set include_localremote parameter
...
modparam("pua_dialoginfo", "include_localremote", 0)
...

1.3.4. caller_confirmed (int)

   Usually the dialog-info of the caller will be "trying -> early
   -> confirmed" and the dialog-info of the callee will be "early
   -> confirmed". On some phones the function LED will start
   blinking if the state is early, regardless if is is the caller
   or the callee (indicated with the "direction" parameter). To
   avoid blinking LEDs for the caller, you can enable this
   parameter. Then the state of the caller will be singaled as
   "confirmed" even in "early" state. This is a workaround for the
   buggy Linksys SPA962 phones. SNOM phones work well with the
   default setting.

   Default value is “0”.

   Example 1.4. Set caller_confirmed parameter
...
modparam("pua_dialoginfo", "caller_confirmed", 1)
...

1.3.5. presence_server (string)

   The address of the presence server, where the PUBLISH messages
   should be sent (not compulsory).

   Example 1.5. Set presence_server parameter
...
modparam("pua_dialoginfo", "presence_server", "sip:ps@opensips.org:5060"
)
...

1.3.6. caller_spec_param (string)

   The name of the pseudovariable that will hold a custom caller
   URI. If this variable is not set, the information in From
   header is used. If you want to use another caller definition,
   you have to fill in this pseudovariable before calling
   dialoginfo_set() function. The format of the string resemples
   the format of To/From SIP headers: "display_name<sip_uri>" or
   "sip_uri".

   Example 1.6. Set caller_spec_param parameter
...
modparam("pua_dialoginfo", "caller_spec_param", "$avp(10)")
...

1.3.7. callee_spec_param (string)

   The name of the pseudovariable that will hold the callee URI.
   If this variable will not be set, the callee information used
   will be made of To display uri + RURI. the. The format of the
   string to set this pseudovariable to is the same as described
   in caller_spec_param section.

   Example 1.7. Set caller_spec_param parameter
...
modparam("pua_dialoginfo", "callee_spec_param", "$avp(11)")
...

1.3.8. osips_ps (int)

   It is advisable to specify if you use a different presence
   server than OpenSIPS presence server, by setting this parameter
   to 0. By default, a trick (version in the Publish body is set
   '0000000') is used when working with Opensips Presence Server
   to make the processing faster and this might not be accepted by
   other presence servers.

   Default value is “1”.

   Example 1.8. Set osips_ps parameter
...
modparam("pua_dialoginfo", "osips_ps", 0)
...

1.4. Exported Functions

1.4.1.  dialoginfo_set([side])

   This function must be called for INVITE messages that
   initialize a dialog for which dialoginfo information must be
   published.

   Meaning of the parameters:
     * side (optional) - can be "A" or "B" for caller or callee
       PUBLISH only - if missing, both sides will be published.

   Example 1.9. dialoginfo_set usage
...
        if(is_method("INVITE"))
                if(uri =~ "opensips.org")
                        dialoginfo_set();
...
