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(Updated Sept. 5, 2014, 6:44 p.m.) Review request for Asterisk Developers. Changes ------- Fixed some reference leaks. Refactored for default processing change in res_phoneprov. Added documentation to pjsip.conf.sample. Repository: Asterisk Description ------- This module allows res_pjsip to integrate with res_phoneprov and depends on the res_phoneprov refactor (r3970). Two new pjsip.conf objects are defined by this module... 'phoneprov_default' which defines defaults for all users created by this provider. [myppdefaults] type=phoneprov_default PROFILE=grandstream ; required SERVER=myserver.example.com OTHERVAR=othervalue 'phoneprov_user' which defines each user to be exposed. [pp1000] type=phoneprov_user endpoint=ep1000 ; optional reference to an existing endpoint MAC=deadbeef4dad ; required PROFILE=grandstream2 ; overrides the default LINEKEYS=1 LINE=1 OTHERVAR=othervalue [pp1001] type=phoneprov_user endpoint=ep1001 ; optional reference to an existing endpoint MAC=deadf00d4dad ; required LINEKEYS=1 LINE=1 LABEL=1001 ; overrides pp1001 OTHERVAR=othervalue USERNAME, CALLERID, DISPLAY_NAME and SECRET are automatically pulled from endpoint and endpoint->auth if defined. LABEL is automatically set from the phoneprov_user id. ENDPOINT_ID, TRANSPORT_ID and AUTH_ID are automatically added. Any other variables defined are automatically passed through and are available for template substitution even if they're not one of the standard variables defined by res_phoneprov. Diffs (updated) ----- branches/12/res/res_pjsip_phoneprov_provider.c PRE-CREATION branches/12/configs/pjsip.conf.sample 422737 Diff: https://reviewboard.asterisk.org/r/3976/diff/ Testing ------- I'm already starting to convert sip peers to pjsip endpoints with no change to my Grandstream templates. Thanks, George Joseph
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