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About.Version-1-6-4 History

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May 14, 2013, at 01:56 PM by razvancrainea -
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!! Main -> [[AvailableVersions|Available Versions]] -> [[About.Version-1-6-x|Version 1.6.x]] -> OpenSIPS Release 1.6.4
to:
!!!!![[About.About | About]] -> [[About.AvailableVersions | Available Versions ]] -> [[About.Version-1-6-x|Version 1.6.x]] -> Release 1.6.4
April 26, 2013, at 04:31 PM by razvancrainea -
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Read the instructions for how to [[Resources.DocsMigration163to164 | migrate from 1.6.3 version to 1.6.4 ]].

to:
Read the instructions for how to [[Documentation.Migration-1-6-3-to-1-6-4 | migrate from 1.6.3 version to 1.6.4 ]].

Changed lines 21-24 from:
** IP - http://www.opensips.org/Resources/DocsCoreTran#toc42
** SDP - http://www.opensips.org/Resources/DocsCoreTran#toc51
** CSV - http://www.opensips.org/Resources/DocsCoreTran#toc48
to:
** [[ Documentation.Script-Tran#toc58 | IP ]]
** [[ Documentation.Script-Tran#toc67 | SDP ]]
** [[ Documentation.Script-Tran#toc64 | CSV ]]
Changed lines 37-38 from:
* added a new core function '''construct_uri()''' which builds a sip uri based on the protocol, username, domain, port and extra params that it receives - http://www.opensips.org/Resources/DocsCoreFcn#toc99
to:
* added a new core function '''construct_uri()''' which builds a sip uri based on the protocol, username, domain, port and extra params that it receives - [[ Documentation.Script-CoreFunctions#toc118 | more info ]]
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* new function '''is_audio_on_hold()'''
to:
* new function '''is_audio_on_hold()'''
April 25, 2013, at 05:20 PM by razvancrainea -
Added lines 1-165:
!! Main -> [[AvailableVersions|Available Versions]] -> [[About.Version-1-6-x|Version 1.6.x]] -> OpenSIPS Release 1.6.4
----
(:toc-float Table of Content:)


----
!!! What is new in 1.6.4 (versus 1.6.3)

Full Changelog can be found [[http://opensips.org/pub/opensips/1.6.4/src/ChangeLog|here]].

----
!!! Migration from 1.6.3 to 1.6.4

Read the instructions for how to [[Resources.DocsMigration163to164 | migrate from 1.6.3 version to 1.6.4 ]].


----
!!!! Core

* new transformations available:
** IP - http://www.opensips.org/Resources/DocsCoreTran#toc42
** SDP - http://www.opensips.org/Resources/DocsCoreTran#toc51
** CSV - http://www.opensips.org/Resources/DocsCoreTran#toc48

* added bitwise operations support for string variables. To be used with the new IP transformations in binary format

* added possibility to set destination domain and destination port from script. You can now set $dd and $dp

* new '''disable_503_translation''' global parameter do disable the 503 -> 500 reply translation. By default the '''disable_503_translation''' is off (translation is done)

* added '''core:timestamp''' new statistic to provide access to the opensips internal timestamp

* added a new pseudovariable called '''argv''', that allows reading arguments specified with '-o' option in command line

* added a new pseudovariable called '''env''', that provides access to environment variables.

* added a new core function '''construct_uri()''' which builds a sip uri based on the protocol, username, domain, port and extra params that it receives - http://www.opensips.org/Resources/DocsCoreFcn#toc99

* added '''Call-Info''' header to the list of known opensips headers

* if '''next_branches()''' returns true, returns 1 if other branches are still pending and return 2 if no other branches are left for future processing - shortly, if 2: this is the last branch, if 1: other will follow

* Reorganizing the core SDP parser:
** sendrecv and ptime are now session attributes
** parse connection and origin IP
** sdp helper logs are now configurable
** parse rtcp attributes
** detect on hold media during sdp parsing



!!!! Tools

* opensipsdbctl : skips grants in case opensips ro and rw database user is root

* opensipsdbctl: allow b2b tables to be installed via opensipsdbctl script


!!!! AAA_RADIUS module

* if type of RADIUS AVP is IPADDR or INTEGER and value is received as string, do the proper conversion when filling in RADIUS packages

* fixed vendor ID when searching radius AVP in radius replies

* expose internal return code for authentication function - you can see in the script if the failure was because of an error or because of auth rejected.


!!!! ACC module

* CDR generation - if dialog support is present, instead of START / STOP events, the ACC module will directly generate a complete CDR (single record per call) : see http://www.opensips.org/html/docs/modules/devel/acc.html#ACC-cdr-id


!!!! AUTH module

* if nonce reused, return "stale" indicater in the challenge


!!!! DIALOG module

* added a new exported function to the dialog module, fix_route_dialog(), which forces an in-dialog SIP message to contain the ruri, route headers and dst_uri, as specified by the internal data of the dialog it belongs to. The function will prevent the existence of bogus injected in-dialog requests ( like malicious BYEs )

* updated the validate_dialog() and fix_route_dialog() exported functions, so that they can now handle all types of routing ( loose to loose, loose to strict, strict to loose and strict to strict )

* instead of error, dialog variables ( $dlg_val() ) return NULL (and not error) if there is no dialog in the current context

* API: added the possibility to terminate a dialog from another module

* included the dialog vals and profiles to be printed as part of dialog context (via dlg_list_ctx MI func)


!!!! DROUTING module

* added '''attributes''' fields for rules - this value is available in script (via AVPs) when the rule is matched and used

* added a new AVP to export the ID of the matched RULE when the rule matches (EX: for accounting purposes)

* added support for NAPTR & SRV lookup (resolving) and multiple IPs when GWs are defined by names


!!!! CFGUTILS mmodule

* '''rand_event()''' may take as optional param (with vars) the probability (this param will override the global probability ,set via set_prob)


!!!! NATHELPER module

* RTP timeout detection and reporting with dialog module integration (call proper call termination on signaling level). See more http://lists.opensips.org/pipermail/users/2010-December/015883.html

* removing obsolete '''force_rtp_proxy()''' function and replaced with new '''rtpproxy_offer()'''/ '''rtpproxy_answer()''' ; obsolete flag 's' removed


* Removing restriction on forcing RTP RELAY only for INVITE type SIP requests. UPDATE is a valid SIP request that can update the SDP.

* '''fix_nated_contact()''' may take as optional param a list of URI params to be added to Contact URI



!!!! PRESENCE module

* added '''bla_fix_remote_target''' parameter that disables a BLA Polycom specific behaviour

* support for extra headers in NOTIFY: - extra headers received in PUBLISH requests can be stored and sent to the subscrber via NOTIFY requests

* fixed a BLA problem - if received a NOTIFY without any dialog received, don't send Notify to the others in the BLA group


!!!! PRESENCE_CALLINFO module (NEW)

The module enables the handling of "call-info" and "line-seize" events inside the presence module. It is used with the general event handling module: presence and it constructs and adds "Call-Info" headers to notification events. To send "call-info" notification to watchers, a third-party application must publish "call-info" events to the presence server.

Read more on http://www.opensips.org/html/docs/modules/devel/presence_callinfo.html


!!!! RR module

* Enhancement to record_route_preset functions to accept a second Record-Route header to be preset via a second argument.

* new module parameter '''enable_socket_mismatch_warning''' - When a preset record-route header is forced in OpenSIPS config and the host from the record-route header is not the same as the host server, a warning will be printed out in the logs. The 'enable_socket_mismatch_warning' parameter enables or disables the warning. When OpenSIPS is behind a NATed firewall, we don't want this warning to be printed for every bridged call.


!!!! TM module

* '''t_replicate''' can now also receive pseudo-variables as argument

* added the possibility to send a reply with body from the script with '''t_reply_with_body''' function

* added new '''t_add_hdrs()''' function to add a custom header to all internally generated requests from a transaction (cancels, acks)


!!!! B2B_LOGIC & B2B_ENTITIES modules

* new: database storage for restart persistence

* new: export API from b2b_logic to allow another controlling layer above it in the form of a module

* new: improved the bridging a party that already is in a call to a new destination so that the user can hear ringing tone until the new destination answers

* fix : reply to both the parties if they send BYE in almost the same time

* fix : include Max-Forwards header in all the requests that require it (Invite, Update, Prack)


!!!! TEXPOPS module

* new function '''is_audio_on_hold()'''

Page last modified on May 14, 2013, at 01:56 PM