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2014-06-16chan_dahdi: Adds support for major update to libss7.Richard Mudgett
* SS7 support now requires libss7 v2.0 or later. The new libss7 is not backwards compatible. * Added SS7 support for connected line and redirecting. * Most SS7 CLI commands are reworked as well as new SS7 commands added. See online CLI help. * Added several SS7 config option parameters described in chan_dahdi.conf.sample. * ISUP timer support reworked and now requires explicit configuration. See ss7.timers.sample. Special thanks to Kaloyan Kovachev for his support and persistence in getting the original patch by adomjan updated and ready for release. SS7-27 #close Reported by: adomjan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-07uniqueid: Fix chan_dahdi, sig_pri, sig_ss7, test_cdr, and test_cel compiler ↵Richard Mudgett
errors. (issue ASTERISK-23120) ........ Merged revisions 410171 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-08Convert sig_ss7 to use a global callback table.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-05Setup DSP when SS7 call is connected or early media is available.Richard Mudgett
Outgoing SS7 calls fail to detect incoming DTMF so any bridged channel that requires out-of-band DTMF will not work. * Added sig_ss7_open_media() calls at appropriate places in sig_ss7.c. The new call converts conditionaled out unconverted code and shows that the code really did something useful. * Improved some chan_dahdi DTMF debug messages to help track DTMF handling. (closes issue ASTERISK-19312) Reported by: Igor Nikolaev ........ Merged revisions 358260 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358261 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-18push 'outgoing' flag from sig_XXX up to chan_dahdiAlec L Davis
'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync. Now provides a callback for all the low level sig_XXX modules. (issue ASTERISK-19316) alecdavis (license 585) Reported by: Jeremy Pepper Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1747/ ........ Merged revisions 355850 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355851 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-01Constify some more channel driver technology callback parameters.Richard Mudgett
Review: https://reviewboard.asterisk.org/r/1707/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-11Add protection for SS7 channel allocation and better glare handling.Richard Mudgett
* Added a CLI "ss7 show channels" command that might prove useful for future debugging. * Made the incoming SS7 channel event check and gripe message uniform. * Made sure that the DNID string for an incoming call is always initialized. (issue ASTERISK-17966) Reported by: Kenneth Van Velthoven Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 340365 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340366 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-08Add private lock deadlock avoidance callback to PRI and SS7.Richard Mudgett
Factor out the equivalent function for analog. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313100 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25Merged revisions 303771 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-14Extract sig_ss7_init_linkset() to sig_ss7.Richard Mudgett
Also found a place where sig_pri_init_pri() was inlined and called it instead. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-09Add missing API function to sig_ss7: sig_ss7_fixup().Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-07Extract sig_ss7 out of chan_dahdi.Richard Mudgett
Extract the SS7 specific code out of chan_dahdi like what was done to ISDN/PRI and analog signaling. The new SS7 structures were modeled on sig_pri. The changes to sig_pri are an enhancement and a bug fix made possible because SS7 was extracted. 1) The sig_pri TRANSFERCAPABILITY channel variable should have been set unconditionally in sig_pri_new_ast_channel(). 2) SS7/PRI transfer capability interaction in dahdi_new() fixed because of SS7 extraction. 3) Module ref count error in dahdi_new() if startpbx failed to start the PBX for some reason. Review: https://reviewboard.asterisk.org/r/661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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