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About.Version-1-6-4 HistoryHide minor edits - Show changes to output May 14, 2013, at 01:56 PM
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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
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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 ]] Changed line 165 from:
* new function '''is_audio_on_hold()''' to:
* new function '''is_audio_on_hold()''' April 25, 2013, at 05:20 PM
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!! 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()''' |