Re: [asterisk-dev] [Code Review] 2963: chan_pjsip: Extend redirect handling support
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/2963/ --- (Updated Nov. 27, 2013, 6:36 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 403207 Repository: Asterisk Description --- chan_pjsip currently supports only one method for handling redirects: It takes the user portion of the target and places it into the call forwarding target as a local extension. This is fine for calling end-user devices but is not suitable for some situations involving other SIP servers (*cough* Microsoft Lync *cough*). The attached patch makes the behavior configurable and adds two other options: uri_dialplan and uri_pjsip. The uri_dialplan option returns the URI as the call forwarding target and instructs the dial process to dial it using the original endpoint. This is the equivalent of the promiscredir option in chan_sip. The uri_pjsip option handles the redirect completely within chan_pjsip itself. This allows multiple targets to be tried if need be, and also reduces the amount of work the core has to do (no channel teardown and dialing again, the same channel is used). As all of these may be useful for people and implementing them is relatively easy I've done so. Diffs - /branches/12/res/res_pjsip_session.c 402863 /branches/12/res/res_pjsip/pjsip_configuration.c 402863 /branches/12/res/res_pjsip.c 402863 /branches/12/include/asterisk/res_pjsip.h 402863 Diff: https://reviewboard.asterisk.org/r/2963/diff/ Testing --- Placed calls to a target with each option, confirmed that they work as expected. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 2963: chan_pjsip: Extend redirect handling support
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/2963/#review10288 --- Ship it! Ship It! - Mark Michelson On Nov. 16, 2013, 4:09 p.m., Joshua Colp wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/2963/ --- (Updated Nov. 16, 2013, 4:09 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- chan_pjsip currently supports only one method for handling redirects: It takes the user portion of the target and places it into the call forwarding target as a local extension. This is fine for calling end-user devices but is not suitable for some situations involving other SIP servers (*cough* Microsoft Lync *cough*). The attached patch makes the behavior configurable and adds two other options: uri_dialplan and uri_pjsip. The uri_dialplan option returns the URI as the call forwarding target and instructs the dial process to dial it using the original endpoint. This is the equivalent of the promiscredir option in chan_sip. The uri_pjsip option handles the redirect completely within chan_pjsip itself. This allows multiple targets to be tried if need be, and also reduces the amount of work the core has to do (no channel teardown and dialing again, the same channel is used). As all of these may be useful for people and implementing them is relatively easy I've done so. Diffs - /branches/12/res/res_pjsip_session.c 402863 /branches/12/res/res_pjsip/pjsip_configuration.c 402863 /branches/12/res/res_pjsip.c 402863 /branches/12/include/asterisk/res_pjsip.h 402863 Diff: https://reviewboard.asterisk.org/r/2963/diff/ Testing --- Placed calls to a target with each option, confirmed that they work as expected. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
