Re: [asterisk-dev] [Code Review] 4327: Various fixes for OS X
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4327/ --- (Updated Jan. 9, 2015, 5:31 p.m.) Review request for Asterisk Developers. Changes --- Fixed compile errors for res_timing_kqueue.c Repository: Asterisk Description --- This patch addresses compilation errors on OS X. It's been a while, so there's quite a few things. * Fixed __attribute__ decls in route.h to be portable. * Fixed htonll and ntohll to work when they are defined as macros. * Replaced sem_t usage with our ast_sem wrapper. * Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC 4.9 warnings using sig*set() functions. * Fixed some format strings for portability. * Added pjproject dependency to res_pjsip_config_wizard.c. Diffs (updated) - /branches/13/res/res_timing_kqueue.c 430428 /branches/13/res/res_pjsip_config_wizard.c 430428 /branches/13/main/sem.c 430428 /branches/13/main/rtp_engine.c 430428 /branches/13/main/bridge_channel.c 430428 /branches/13/main/asterisk.c 430428 /branches/13/main/app.c 430428 /branches/13/include/asterisk/sem.h 430428 /branches/13/include/asterisk/autoconfig.h.in 430428 /branches/13/funcs/func_presencestate.c 430428 /branches/13/configure.ac 430428 /branches/13/configure UNKNOWN /branches/13/channels/sip/include/route.h 430428 Diff: https://reviewboard.asterisk.org/r/4327/diff/ Testing --- Compiled on both OS X and Linux. Thanks, David Lee -- _ -- 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] 4327: Various fixes for OS X
On Jan. 9, 2015, 11:16 a.m., George Joseph wrote: There are a few recent issues that I think this addresses... ASTERISK-24539 ASTERISK-24544 Does this patch also address ASTERISK-24559 ASTERISK-24565 ? David Lee wrote: ASTERISK-24539 ASTERISK-24544 Yes. ASTERISK-24559 Probably not. ASTERISK-24565 No. George Joseph wrote: Ok, can you add the 2 yeses to the review? For ASTERISK-24559 (app_voicemail notify_new_message() forked process crashes on OS X), that beyond my current scope. For ASTERISK-24565 (kqueue not working correctly on OSX), I did fix the compile errors, but it still doesn't function properly. I spent a few minutes looking at it, but it looks pretty fundamentally broken to me. (As in the kqueue code there doesn't line up with kqueue timer examples that I see.) I wouldn't know how to get it to work without learning a lot more about kqueue than I care to learn. FWIW, it appears in r271657, there was an attempt to disable kqueue on OS X anyways, but the conflict statement that was introduced doesn't seem to be doing that anyways. I may try to fix that, so in the very least no one tries to use it on OS X. - David --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4327/#review14151 --- On Jan. 9, 2015, 5:31 p.m., David Lee wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4327/ --- (Updated Jan. 9, 2015, 5:31 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- This patch addresses compilation errors on OS X. It's been a while, so there's quite a few things. * Fixed __attribute__ decls in route.h to be portable. * Fixed htonll and ntohll to work when they are defined as macros. * Replaced sem_t usage with our ast_sem wrapper. * Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC 4.9 warnings using sig*set() functions. * Fixed some format strings for portability. * Added pjproject dependency to res_pjsip_config_wizard.c. Diffs - /branches/13/res/res_timing_kqueue.c 430428 /branches/13/res/res_pjsip_config_wizard.c 430428 /branches/13/main/sem.c 430428 /branches/13/main/rtp_engine.c 430428 /branches/13/main/bridge_channel.c 430428 /branches/13/main/asterisk.c 430428 /branches/13/main/app.c 430428 /branches/13/include/asterisk/sem.h 430428 /branches/13/include/asterisk/autoconfig.h.in 430428 /branches/13/funcs/func_presencestate.c 430428 /branches/13/configure.ac 430428 /branches/13/configure UNKNOWN /branches/13/channels/sip/include/route.h 430428 Diff: https://reviewboard.asterisk.org/r/4327/diff/ Testing --- Compiled on both OS X and Linux. Thanks, David Lee -- _ -- 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] 4311: res_pjsip: make it unloadable
On Jan. 8, 2015, 2:13 p.m., Mark Michelson wrote: branches/13/res/res_pjsip.c, lines 43-45 https://reviewboard.asterisk.org/r/4311/diff/1-2/?file=70130#file70130line43 What caused these dependencies to be added? Kevin Harwell wrote: pjsip_options uses sorcery_memory with regards to qualify/contact_status and contacts themselves are stored in the astdb (location.c) Joshua Colp wrote: I think res_sorcery_memory is fine being a dependency because it is actually required by stuff internally. res_sorcery_astdb on the other hand is only required if you are using the default of storing within astdb - it's runtime configurable to be different. astdb is the persistent store for contacts. res_pjsip will fail to load if res_sorcery_astdb isn't loaded. - George --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4311/#review14126 --- On Jan. 9, 2015, 8:37 a.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4311/ --- (Updated Jan. 9, 2015, 8:37 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24485 https://issues.asterisk.org/jira/browse/ASTERISK-24485 Repository: Asterisk Description --- The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue by Corey Farrell with a few modifications. Removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. Diffs - branches/13/res/res_pjsip/pjsip_outbound_auth.c 430294 branches/13/res/res_pjsip/pjsip_options.c 430294 branches/13/res/res_pjsip/pjsip_global_headers.c 430294 branches/13/res/res_pjsip/pjsip_distributor.c 430294 branches/13/res/res_pjsip/pjsip_configuration.c 430294 branches/13/res/res_pjsip/location.c 430294 branches/13/res/res_pjsip/include/res_pjsip_private.h 430294 branches/13/res/res_pjsip/config_transport.c 430294 branches/13/res/res_pjsip/config_auth.c 430294 branches/13/res/res_pjsip.c 430294 branches/13/main/stasis_message_router.c 430294 Diff: https://reviewboard.asterisk.org/r/4311/diff/ Testing --- Made it so res_pjsip was the only pjsip module loaded and then issued an unload and noted it unloaded successfully (also loaded/unloaded it several times from the CLI). Also when loaded and with REF_DEBUG enabled issued a core stop gracefully and made sure there were no ref leaks for the module. Also tested unloading with other dependent pjsip modules loaded and noted that the module would not unload (as it should since dependencies are currently loaded). And then shutdown asterisk and made sure it did not crash or anything. Started asterisk with nominal and off nominal module and pjsip configurations to make sure things behaved appropriately (no crashes and such) and then attempted to, or successfully unload the res_pjsip module. Also made sure Asterisk continued to shutdown without incident. Thanks, Kevin Harwell -- _ -- 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] 4311: res_pjsip: make it unloadable
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4311/ --- (Updated Jan. 9, 2015, 9:37 a.m.) Review request for Asterisk Developers. Changes --- Removed the MODULEINFO dependency for ast_db Bugs: ASTERISK-24485 https://issues.asterisk.org/jira/browse/ASTERISK-24485 Repository: Asterisk Description --- The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue by Corey Farrell with a few modifications. Removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. Diffs (updated) - branches/13/res/res_pjsip/pjsip_outbound_auth.c 430294 branches/13/res/res_pjsip/pjsip_options.c 430294 branches/13/res/res_pjsip/pjsip_global_headers.c 430294 branches/13/res/res_pjsip/pjsip_distributor.c 430294 branches/13/res/res_pjsip/pjsip_configuration.c 430294 branches/13/res/res_pjsip/location.c 430294 branches/13/res/res_pjsip/include/res_pjsip_private.h 430294 branches/13/res/res_pjsip/config_transport.c 430294 branches/13/res/res_pjsip/config_auth.c 430294 branches/13/res/res_pjsip.c 430294 branches/13/main/stasis_message_router.c 430294 Diff: https://reviewboard.asterisk.org/r/4311/diff/ Testing --- Made it so res_pjsip was the only pjsip module loaded and then issued an unload and noted it unloaded successfully (also loaded/unloaded it several times from the CLI). Also when loaded and with REF_DEBUG enabled issued a core stop gracefully and made sure there were no ref leaks for the module. Also tested unloading with other dependent pjsip modules loaded and noted that the module would not unload (as it should since dependencies are currently loaded). And then shutdown asterisk and made sure it did not crash or anything. Started asterisk with nominal and off nominal module and pjsip configurations to make sure things behaved appropriately (no crashes and such) and then attempted to, or successfully unload the res_pjsip module. Also made sure Asterisk continued to shutdown without incident. Thanks, Kevin Harwell -- _ -- 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] 4325: PJSIP: Prevent slow graceful shutdown when outbound publications have not started
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4325/ --- (Updated Jan. 9, 2015, 3:45 p.m.) Review request for Asterisk Developers and Kevin Harwell. Changes --- Switch to explicit_publish_destroy() instead of kill_it() Bugs: ASTERISK-24655 https://issues.asterisk.org/jira/browse/ASTERISK-24655 Repository: Asterisk Description --- If an outbound publish is configured, and that publication never published any data, then a graceful shutdown would result in Asterisk hanging for a while before finally shutting down. The reason is that the code did not take into account the case where we never started publishing anything. The code would attempt to send a PUBLISH to stop publication, relying on the PUBLISH callback to be called so we could then destroy the PJSIP publishc structure, destroy our publication client, and signal to the unloading code that we were done. The problem is that the PUBLISH callback was never being called, presumably since pjsip_publishc_send() was failing. I modified the code to outright destroy the PJSIP publishc structure and drop its reference to our publication client if we never actually started publishing anything. This way, shutting down gracefully doesn't wait for a callback that will never occur. Also, feel free to give a better suggestion for a function name than kill_it. There are already so many variations on destroying clients in that file, I was a bit bankrupt for ideas. Diffs (updated) - /branches/13/res/res_pjsip_outbound_publish.c 430354 Diff: https://reviewboard.asterisk.org/r/4325/diff/ Testing --- Reproduced the issue originally as described in ASTERISK-24655. With the patch here, I have confirmed that the issue no longer occurs. I won't be writing a testsuite test for this, since the testsuite is not really well suited to this sort of thing. Thanks, Mark Michelson -- _ -- 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] 4325: PJSIP: Prevent slow graceful shutdown when outbound publications have not started
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4325/#review14145 --- Ship it! Ship It! - Joshua Colp On Jan. 9, 2015, 3:45 p.m., Mark Michelson wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4325/ --- (Updated Jan. 9, 2015, 3:45 p.m.) Review request for Asterisk Developers and Kevin Harwell. Bugs: ASTERISK-24655 https://issues.asterisk.org/jira/browse/ASTERISK-24655 Repository: Asterisk Description --- If an outbound publish is configured, and that publication never published any data, then a graceful shutdown would result in Asterisk hanging for a while before finally shutting down. The reason is that the code did not take into account the case where we never started publishing anything. The code would attempt to send a PUBLISH to stop publication, relying on the PUBLISH callback to be called so we could then destroy the PJSIP publishc structure, destroy our publication client, and signal to the unloading code that we were done. The problem is that the PUBLISH callback was never being called, presumably since pjsip_publishc_send() was failing. I modified the code to outright destroy the PJSIP publishc structure and drop its reference to our publication client if we never actually started publishing anything. This way, shutting down gracefully doesn't wait for a callback that will never occur. Also, feel free to give a better suggestion for a function name than kill_it. There are already so many variations on destroying clients in that file, I was a bit bankrupt for ideas. Diffs - /branches/13/res/res_pjsip_outbound_publish.c 430354 Diff: https://reviewboard.asterisk.org/r/4325/diff/ Testing --- Reproduced the issue originally as described in ASTERISK-24655. With the patch here, I have confirmed that the issue no longer occurs. I won't be writing a testsuite test for this, since the testsuite is not really well suited to this sort of thing. Thanks, Mark Michelson -- _ -- 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] 4315: AMI: Make AMI actions that generate event lists consistent.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4315/ --- (Updated Jan. 9, 2015, 10:30 a.m.) Review request for Asterisk Developers. Changes --- Restore the diff for this review and fix the CHANGES note issue. Bugs: ASTERISK-24049 https://issues.asterisk.org/jira/browse/ASTERISK-24049 Repository: Asterisk Description --- * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to Start so the start capitalization is consistent. i.e., The FAXSessions used Start while the rest of the system used start. The corresponding complete event always used Complete. * Fixed ami_show_resource_lists() PJSIPShowResourceLists to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as Header: text. * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). Diffs (updated) - /branches/13/res/res_pjsip_registrar.c 430433 /branches/13/res/res_pjsip_pubsub.c 430433 /branches/13/res/res_pjsip_outbound_registration.c 430433 /branches/13/res/res_pjsip/pjsip_configuration.c 430433 /branches/13/res/res_mwi_external_ami.c 430433 /branches/13/res/res_manager_presencestate.c 430433 /branches/13/res/res_manager_devicestate.c 430433 /branches/13/res/res_fax.c 430433 /branches/13/res/parking/parking_manager.c 430433 /branches/13/main/pbx.c 430433 /branches/13/main/manager_bridges.c 430433 /branches/13/main/manager.c 430433 /branches/13/main/db.c 430433 /branches/13/main/bridge.c 430433 /branches/13/include/asterisk/manager.h 430433 /branches/13/channels/chan_skinny.c 430433 /branches/13/channels/chan_sip.c 430433 /branches/13/channels/chan_iax2.c 430433 /branches/13/channels/chan_dahdi.c 430433 /branches/13/apps/app_voicemail.c 430433 /branches/13/apps/app_queue.c 430433 /branches/13/apps/app_meetme.c 430433 /branches/13/apps/app_confbridge.c 430433 /branches/13/apps/app_agent_pool.c 430433 /branches/13/UPGRADE.txt 430433 /branches/13/CHANGES 430433 Diff: https://reviewboard.asterisk.org/r/4315/diff/ Testing --- Issued all of the AMI actions listed above to verify that the output was consistent. Thanks, rmudgett -- _ -- 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] 4325: PJSIP: Prevent slow graceful shutdown when outbound publications have not started
On Jan. 9, 2015, 2:13 p.m., Joshua Colp wrote: /branches/13/res/res_pjsip_outbound_publish.c, line 665 https://reviewboard.asterisk.org/r/4325/diff/1/?file=70346#file70346line665 explicit_publish_destroy? Good enough for me! - Mark --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4325/#review14138 --- On Jan. 9, 2015, 12:07 a.m., Mark Michelson wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4325/ --- (Updated Jan. 9, 2015, 12:07 a.m.) Review request for Asterisk Developers and Kevin Harwell. Bugs: ASTERISK-24655 https://issues.asterisk.org/jira/browse/ASTERISK-24655 Repository: Asterisk Description --- If an outbound publish is configured, and that publication never published any data, then a graceful shutdown would result in Asterisk hanging for a while before finally shutting down. The reason is that the code did not take into account the case where we never started publishing anything. The code would attempt to send a PUBLISH to stop publication, relying on the PUBLISH callback to be called so we could then destroy the PJSIP publishc structure, destroy our publication client, and signal to the unloading code that we were done. The problem is that the PUBLISH callback was never being called, presumably since pjsip_publishc_send() was failing. I modified the code to outright destroy the PJSIP publishc structure and drop its reference to our publication client if we never actually started publishing anything. This way, shutting down gracefully doesn't wait for a callback that will never occur. Also, feel free to give a better suggestion for a function name than kill_it. There are already so many variations on destroying clients in that file, I was a bit bankrupt for ideas. Diffs - /branches/13/res/res_pjsip_outbound_publish.c 430372 Diff: https://reviewboard.asterisk.org/r/4325/diff/ Testing --- Reproduced the issue originally as described in ASTERISK-24655. With the patch here, I have confirmed that the issue no longer occurs. I won't be writing a testsuite test for this, since the testsuite is not really well suited to this sort of thing. Thanks, Mark Michelson -- _ -- 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] 4311: res_pjsip: make it unloadable
On Jan. 8, 2015, 9:13 p.m., Mark Michelson wrote: branches/13/res/res_pjsip.c, lines 43-45 https://reviewboard.asterisk.org/r/4311/diff/1-2/?file=70130#file70130line43 What caused these dependencies to be added? Kevin Harwell wrote: pjsip_options uses sorcery_memory with regards to qualify/contact_status and contacts themselves are stored in the astdb (location.c) Joshua Colp wrote: I think res_sorcery_memory is fine being a dependency because it is actually required by stuff internally. res_sorcery_astdb on the other hand is only required if you are using the default of storing within astdb - it's runtime configurable to be different. George Joseph wrote: astdb is the persistent store for contacts. res_pjsip will fail to load if res_sorcery_astdb isn't loaded. In a default configuration. - Joshua --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4311/#review14126 --- On Jan. 9, 2015, 3:37 p.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4311/ --- (Updated Jan. 9, 2015, 3:37 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24485 https://issues.asterisk.org/jira/browse/ASTERISK-24485 Repository: Asterisk Description --- The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue by Corey Farrell with a few modifications. Removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. Diffs - branches/13/res/res_pjsip/pjsip_outbound_auth.c 430294 branches/13/res/res_pjsip/pjsip_options.c 430294 branches/13/res/res_pjsip/pjsip_global_headers.c 430294 branches/13/res/res_pjsip/pjsip_distributor.c 430294 branches/13/res/res_pjsip/pjsip_configuration.c 430294 branches/13/res/res_pjsip/location.c 430294 branches/13/res/res_pjsip/include/res_pjsip_private.h 430294 branches/13/res/res_pjsip/config_transport.c 430294 branches/13/res/res_pjsip/config_auth.c 430294 branches/13/res/res_pjsip.c 430294 branches/13/main/stasis_message_router.c 430294 Diff: https://reviewboard.asterisk.org/r/4311/diff/ Testing --- Made it so res_pjsip was the only pjsip module loaded and then issued an unload and noted it unloaded successfully (also loaded/unloaded it several times from the CLI). Also when loaded and with REF_DEBUG enabled issued a core stop gracefully and made sure there were no ref leaks for the module. Also tested unloading with other dependent pjsip modules loaded and noted that the module would not unload (as it should since dependencies are currently loaded). And then shutdown asterisk and made sure it did not crash or anything. Started asterisk with nominal and off nominal module and pjsip configurations to make sure things behaved appropriately (no crashes and such) and then attempted to, or successfully unload the res_pjsip module. Also made sure Asterisk continued to shutdown without incident. Thanks, Kevin Harwell -- _ -- 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] 4315: AMI: Make AMI actions that generate event lists consistent.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4315/ --- (Updated Jan. 9, 2015, 9:51 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24049 https://issues.asterisk.org/jira/browse/ASTERISK-24049 Repository: Asterisk Description --- * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to Start so the start capitalization is consistent. i.e., The FAXSessions used Start while the rest of the system used start. The corresponding complete event always used Complete. * Fixed ami_show_resource_lists() PJSIPShowResourceLists to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as Header: text. * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). Diffs (updated) - /branches/13/res/res_pjsip_outbound_publish.c 430354 Diff: https://reviewboard.asterisk.org/r/4315/diff/ Testing --- Issued all of the AMI actions listed above to verify that the output was consistent. Thanks, rmudgett -- _ -- 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] 4311: res_pjsip: make it unloadable
On Jan. 8, 2015, 2:13 p.m., Mark Michelson wrote: branches/13/res/res_pjsip.c, lines 43-45 https://reviewboard.asterisk.org/r/4311/diff/1-2/?file=70130#file70130line43 What caused these dependencies to be added? Kevin Harwell wrote: pjsip_options uses sorcery_memory with regards to qualify/contact_status and contacts themselves are stored in the astdb (location.c) Joshua Colp wrote: I think res_sorcery_memory is fine being a dependency because it is actually required by stuff internally. res_sorcery_astdb on the other hand is only required if you are using the default of storing within astdb - it's runtime configurable to be different. George Joseph wrote: astdb is the persistent store for contacts. res_pjsip will fail to load if res_sorcery_astdb isn't loaded. Joshua Colp wrote: In a default configuration. True but my point in asking for MODULEINFO to be updated was that it's not obvious that res_pjsip needs astdb to load in most cases. In fact, if you're setting up realtime what would even trigger you to override contact since contact isn't an object you normally define in pjsip.conf? So unless you actually looked in location.c, contact is going to need astdb. - George --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4311/#review14126 --- On Jan. 9, 2015, 8:37 a.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4311/ --- (Updated Jan. 9, 2015, 8:37 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24485 https://issues.asterisk.org/jira/browse/ASTERISK-24485 Repository: Asterisk Description --- The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue by Corey Farrell with a few modifications. Removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. Diffs - branches/13/res/res_pjsip/pjsip_outbound_auth.c 430294 branches/13/res/res_pjsip/pjsip_options.c 430294 branches/13/res/res_pjsip/pjsip_global_headers.c 430294 branches/13/res/res_pjsip/pjsip_distributor.c 430294 branches/13/res/res_pjsip/pjsip_configuration.c 430294 branches/13/res/res_pjsip/location.c 430294 branches/13/res/res_pjsip/include/res_pjsip_private.h 430294 branches/13/res/res_pjsip/config_transport.c 430294 branches/13/res/res_pjsip/config_auth.c 430294 branches/13/res/res_pjsip.c 430294 branches/13/main/stasis_message_router.c 430294 Diff: https://reviewboard.asterisk.org/r/4311/diff/ Testing --- Made it so res_pjsip was the only pjsip module loaded and then issued an unload and noted it unloaded successfully (also loaded/unloaded it several times from the CLI). Also when loaded and with REF_DEBUG enabled issued a core stop gracefully and made sure there were no ref leaks for the module. Also tested unloading with other dependent pjsip modules loaded and noted that the module would not unload (as it should since dependencies are currently loaded). And then shutdown asterisk and made sure it did not crash or anything. Started asterisk with nominal and off nominal module and pjsip configurations to make sure things behaved appropriately (no crashes and such) and then attempted to, or successfully unload the res_pjsip module. Also made sure Asterisk continued to shutdown without incident. Thanks, Kevin Harwell -- _ -- 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] 4315: AMI: Make AMI actions that generate event lists consistent.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4315/#review14147 --- This is not the correct diff for this review. - rmudgett On Jan. 9, 2015, 10:30 a.m., rmudgett wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4315/ --- (Updated Jan. 9, 2015, 10:30 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24049 https://issues.asterisk.org/jira/browse/ASTERISK-24049 Repository: Asterisk Description --- * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to Start so the start capitalization is consistent. i.e., The FAXSessions used Start while the rest of the system used start. The corresponding complete event always used Complete. * Fixed ami_show_resource_lists() PJSIPShowResourceLists to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as Header: text. * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). Diffs - /branches/13/res/res_pjsip_registrar.c 430433 /branches/13/res/res_pjsip_pubsub.c 430433 /branches/13/res/res_pjsip_outbound_registration.c 430433 /branches/13/res/res_pjsip/pjsip_configuration.c 430433 /branches/13/res/res_mwi_external_ami.c 430433 /branches/13/res/res_manager_presencestate.c 430433 /branches/13/res/res_manager_devicestate.c 430433 /branches/13/res/res_fax.c 430433 /branches/13/res/parking/parking_manager.c 430433 /branches/13/main/pbx.c 430433 /branches/13/main/manager_bridges.c 430433 /branches/13/main/manager.c 430433 /branches/13/main/db.c 430433 /branches/13/main/bridge.c 430433 /branches/13/include/asterisk/manager.h 430433 /branches/13/channels/chan_skinny.c 430433 /branches/13/channels/chan_sip.c 430433 /branches/13/channels/chan_iax2.c 430433 /branches/13/channels/chan_dahdi.c 430433 /branches/13/apps/app_voicemail.c 430433 /branches/13/apps/app_queue.c 430433 /branches/13/apps/app_meetme.c 430433 /branches/13/apps/app_confbridge.c 430433 /branches/13/apps/app_agent_pool.c 430433 /branches/13/UPGRADE.txt 430433 /branches/13/CHANGES 430433 Diff: https://reviewboard.asterisk.org/r/4315/diff/ Testing --- Issued all of the AMI actions listed above to verify that the output was consistent. Thanks, rmudgett -- _ -- 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] 4327: Various fixes for OS X
On Jan. 9, 2015, 10:16 a.m., George Joseph wrote: There are a few recent issues that I think this addresses... ASTERISK-24539 ASTERISK-24544 Does this patch also address ASTERISK-24559 ASTERISK-24565 ? David Lee wrote: ASTERISK-24539 ASTERISK-24544 Yes. ASTERISK-24559 Probably not. ASTERISK-24565 No. Ok, can you add the 2 yeses to the review? - George --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4327/#review14151 --- On Jan. 9, 2015, 10:05 a.m., David Lee wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4327/ --- (Updated Jan. 9, 2015, 10:05 a.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- This patch addresses compilation errors on OS X. It's been a while, so there's quite a few things. * Fixed __attribute__ decls in route.h to be portable. * Fixed htonll and ntohll to work when they are defined as macros. * Replaced sem_t usage with our ast_sem wrapper. * Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC 4.9 warnings using sig*set() functions. * Fixed some format strings for portability. * Added pjproject dependency to res_pjsip_config_wizard.c. Diffs - /branches/13/res/res_pjsip_config_wizard.c 430428 /branches/13/main/sem.c 430428 /branches/13/main/rtp_engine.c 430428 /branches/13/main/bridge_channel.c 430428 /branches/13/main/asterisk.c 430428 /branches/13/main/app.c 430428 /branches/13/include/asterisk/sem.h 430428 /branches/13/include/asterisk/autoconfig.h.in 430428 /branches/13/funcs/func_presencestate.c 430428 /branches/13/configure.ac 430428 /branches/13/configure UNKNOWN /branches/13/channels/sip/include/route.h 430428 Diff: https://reviewboard.asterisk.org/r/4327/diff/ Testing --- Compiled on both OS X and Linux. Thanks, David Lee -- _ -- 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] 4327: Various fixes for OS X
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4327/#review14151 --- There are a few recent issues that I think this addresses... ASTERISK-24539 ASTERISK-24544 Does this patch also address ASTERISK-24559 ASTERISK-24565 ? - George Joseph On Jan. 9, 2015, 10:05 a.m., David Lee wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4327/ --- (Updated Jan. 9, 2015, 10:05 a.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- This patch addresses compilation errors on OS X. It's been a while, so there's quite a few things. * Fixed __attribute__ decls in route.h to be portable. * Fixed htonll and ntohll to work when they are defined as macros. * Replaced sem_t usage with our ast_sem wrapper. * Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC 4.9 warnings using sig*set() functions. * Fixed some format strings for portability. * Added pjproject dependency to res_pjsip_config_wizard.c. Diffs - /branches/13/res/res_pjsip_config_wizard.c 430428 /branches/13/main/sem.c 430428 /branches/13/main/rtp_engine.c 430428 /branches/13/main/bridge_channel.c 430428 /branches/13/main/asterisk.c 430428 /branches/13/main/app.c 430428 /branches/13/include/asterisk/sem.h 430428 /branches/13/include/asterisk/autoconfig.h.in 430428 /branches/13/funcs/func_presencestate.c 430428 /branches/13/configure.ac 430428 /branches/13/configure UNKNOWN /branches/13/channels/sip/include/route.h 430428 Diff: https://reviewboard.asterisk.org/r/4327/diff/ Testing --- Compiled on both OS X and Linux. Thanks, David Lee -- _ -- 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] 4327: Various fixes for OS X
On Jan. 9, 2015, 11:16 a.m., George Joseph wrote: There are a few recent issues that I think this addresses... ASTERISK-24539 ASTERISK-24544 Does this patch also address ASTERISK-24559 ASTERISK-24565 ? ASTERISK-24539 ASTERISK-24544 Yes. ASTERISK-24559 Probably not. ASTERISK-24565 No. - David --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4327/#review14151 --- On Jan. 9, 2015, 11:05 a.m., David Lee wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4327/ --- (Updated Jan. 9, 2015, 11:05 a.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- This patch addresses compilation errors on OS X. It's been a while, so there's quite a few things. * Fixed __attribute__ decls in route.h to be portable. * Fixed htonll and ntohll to work when they are defined as macros. * Replaced sem_t usage with our ast_sem wrapper. * Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC 4.9 warnings using sig*set() functions. * Fixed some format strings for portability. * Added pjproject dependency to res_pjsip_config_wizard.c. Diffs - /branches/13/res/res_pjsip_config_wizard.c 430428 /branches/13/main/sem.c 430428 /branches/13/main/rtp_engine.c 430428 /branches/13/main/bridge_channel.c 430428 /branches/13/main/asterisk.c 430428 /branches/13/main/app.c 430428 /branches/13/include/asterisk/sem.h 430428 /branches/13/include/asterisk/autoconfig.h.in 430428 /branches/13/funcs/func_presencestate.c 430428 /branches/13/configure.ac 430428 /branches/13/configure UNKNOWN /branches/13/channels/sip/include/route.h 430428 Diff: https://reviewboard.asterisk.org/r/4327/diff/ Testing --- Compiled on both OS X and Linux. Thanks, David Lee -- _ -- 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] 4315: AMI: Make AMI actions that generate event lists consistent.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4315/ --- (Updated Jan. 9, 2015, 11:55 a.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 430434 Bugs: ASTERISK-24049 https://issues.asterisk.org/jira/browse/ASTERISK-24049 Repository: Asterisk Description --- * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to Start so the start capitalization is consistent. i.e., The FAXSessions used Start while the rest of the system used start. The corresponding complete event always used Complete. * Fixed ami_show_resource_lists() PJSIPShowResourceLists to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as Header: text. * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). Diffs - /branches/13/res/res_pjsip_registrar.c 430433 /branches/13/res/res_pjsip_pubsub.c 430433 /branches/13/res/res_pjsip_outbound_registration.c 430433 /branches/13/res/res_pjsip/pjsip_configuration.c 430433 /branches/13/res/res_mwi_external_ami.c 430433 /branches/13/res/res_manager_presencestate.c 430433 /branches/13/res/res_manager_devicestate.c 430433 /branches/13/res/res_fax.c 430433 /branches/13/res/parking/parking_manager.c 430433 /branches/13/main/pbx.c 430433 /branches/13/main/manager_bridges.c 430433 /branches/13/main/manager.c 430433 /branches/13/main/db.c 430433 /branches/13/main/bridge.c 430433 /branches/13/include/asterisk/manager.h 430433 /branches/13/channels/chan_skinny.c 430433 /branches/13/channels/chan_sip.c 430433 /branches/13/channels/chan_iax2.c 430433 /branches/13/channels/chan_dahdi.c 430433 /branches/13/apps/app_voicemail.c 430433 /branches/13/apps/app_queue.c 430433 /branches/13/apps/app_meetme.c 430433 /branches/13/apps/app_confbridge.c 430433 /branches/13/apps/app_agent_pool.c 430433 /branches/13/UPGRADE.txt 430433 /branches/13/CHANGES 430433 Diff: https://reviewboard.asterisk.org/r/4315/diff/ Testing --- Issued all of the AMI actions listed above to verify that the output was consistent. Thanks, rmudgett -- _ -- 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] 4315: AMI: Make AMI actions that generate event lists consistent.
On Jan. 9, 2015, 10:31 a.m., rmudgett wrote: This is not the correct diff for this review. Diff 3 that is. - rmudgett --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4315/#review14147 --- On Jan. 9, 2015, 10:30 a.m., rmudgett wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4315/ --- (Updated Jan. 9, 2015, 10:30 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24049 https://issues.asterisk.org/jira/browse/ASTERISK-24049 Repository: Asterisk Description --- * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to Start so the start capitalization is consistent. i.e., The FAXSessions used Start while the rest of the system used start. The corresponding complete event always used Complete. * Fixed ami_show_resource_lists() PJSIPShowResourceLists to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as Header: text. * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). Diffs - /branches/13/res/res_pjsip_registrar.c 430433 /branches/13/res/res_pjsip_pubsub.c 430433 /branches/13/res/res_pjsip_outbound_registration.c 430433 /branches/13/res/res_pjsip/pjsip_configuration.c 430433 /branches/13/res/res_mwi_external_ami.c 430433 /branches/13/res/res_manager_presencestate.c 430433 /branches/13/res/res_manager_devicestate.c 430433 /branches/13/res/res_fax.c 430433 /branches/13/res/parking/parking_manager.c 430433 /branches/13/main/pbx.c 430433 /branches/13/main/manager_bridges.c 430433 /branches/13/main/manager.c 430433 /branches/13/main/db.c 430433 /branches/13/main/bridge.c 430433 /branches/13/include/asterisk/manager.h 430433 /branches/13/channels/chan_skinny.c 430433 /branches/13/channels/chan_sip.c 430433 /branches/13/channels/chan_iax2.c 430433 /branches/13/channels/chan_dahdi.c 430433 /branches/13/apps/app_voicemail.c 430433 /branches/13/apps/app_queue.c 430433 /branches/13/apps/app_meetme.c 430433 /branches/13/apps/app_confbridge.c 430433 /branches/13/apps/app_agent_pool.c 430433 /branches/13/UPGRADE.txt 430433 /branches/13/CHANGES 430433 Diff: https://reviewboard.asterisk.org/r/4315/diff/ Testing --- Issued all of the AMI actions listed above to verify that the output was consistent. Thanks, rmudgett -- _ -- 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
[asterisk-dev] [Code Review] 4327: Various fixes for OS X
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4327/ --- Review request for Asterisk Developers. Repository: Asterisk Description --- This patch addresses compilation errors on OS X. It's been a while, so there's quite a few things. * Fixed __attribute__ decls in route.h to be portable. * Fixed htonll and ntohll to work when they are defined as macros. * Replaced sem_t usage with our ast_sem wrapper. * Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC 4.9 warnings using sig*set() functions. * Fixed some format strings for portability. * Added pjproject dependency to res_pjsip_config_wizard.c. Diffs - /branches/13/res/res_pjsip_config_wizard.c 430428 /branches/13/main/sem.c 430428 /branches/13/main/rtp_engine.c 430428 /branches/13/main/bridge_channel.c 430428 /branches/13/main/asterisk.c 430428 /branches/13/main/app.c 430428 /branches/13/include/asterisk/sem.h 430428 /branches/13/include/asterisk/autoconfig.h.in 430428 /branches/13/funcs/func_presencestate.c 430428 /branches/13/configure.ac 430428 /branches/13/configure UNKNOWN /branches/13/channels/sip/include/route.h 430428 Diff: https://reviewboard.asterisk.org/r/4327/diff/ Testing --- Compiled on both OS X and Linux. Thanks, David Lee -- _ -- 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] 4267: Testsuite: Add blind transfer tests for Stasis application interaction.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4267/ --- (Updated Jan. 9, 2015, 11:45 a.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers and Mark Michelson. Changes --- Committed in revision 6220 Bugs: ASTERISK-24581 and ASTERISK-24649 https://issues.asterisk.org/jira/browse/ASTERISK-24581 https://issues.asterisk.org/jira/browse/ASTERISK-24649 Repository: testsuite Description --- This adds the remaining blind transfer tests 1.9 1.10 as described on the StasisStart/StasisEnd Test Plan at: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30279826 This additionally updates the existing test 'stasis_bridge_to_non_stasis_app' (1.8) to verify the StasisEnd events of the channels per the test plan. An additional (dummy) channel was added for the test to prevent the test from ending when the channels involved in the test are hung up. This allows the StasisEnd events of all the other channels to be verified before the test has ended. The test description has also been updated to include more details about the test. The two new tests use the 'call_transfer.py' module which is a modified copy of tests/channels/pjsip/transfers/blind_transfer/caller_refer_only/transfer.py for these two new tests. The module uses the pjsua python library to place calls into Asterisk and perform the blind transfer. Notes: * Due to the architecture of pjsua_mod.py, the call_transfer.py module is used as both a pluggable module and a callback module. With a little more work the module could be made to handle other common variations (ex. who places calls, who receives a call, who performs the transfer, handle a transfer target that is another pjsua endpoint like the original module) with everything configurable via YAML. I imagine it would be useful for future tests. Any takers? :) * The bug ASTERISK-24649 was found during the development of the new tests here and will likely cause them to fail every so often. Diffs - /asterisk/trunk/tests/rest_api/external_interaction/blind_transfer/tests.yaml 6155 /asterisk/trunk/tests/rest_api/external_interaction/blind_transfer/stasis_bridge_to_same_stasis_app/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/blind_transfer/stasis_bridge_to_same_stasis_app/configs/ast1/pjsip.conf PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/blind_transfer/stasis_bridge_to_same_stasis_app/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/blind_transfer/stasis_bridge_to_non_stasis_app/test-config.yaml 6155 /asterisk/trunk/tests/rest_api/external_interaction/blind_transfer/stasis_bridge_to_non_stasis_app/configs/ast1/extensions.conf 6155 /asterisk/trunk/tests/rest_api/external_interaction/blind_transfer/stasis_bridge_to_non_stasis_app/blind_transfer.py 6155 /asterisk/trunk/tests/rest_api/external_interaction/blind_transfer/stasis_bridge_to_different_stasis_app/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/blind_transfer/stasis_bridge_to_different_stasis_app/configs/ast1/pjsip.conf PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/blind_transfer/stasis_bridge_to_different_stasis_app/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/blind_transfer/call_transfer.py PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4267/diff/ Testing --- * Executed tests multiple times * Reviewed logs to manually verify StasisStart/StasisEnd events occurred. Thanks, jbigelow -- _ -- 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] 4311: res_pjsip: make it unloadable
On Jan. 8, 2015, 9:13 p.m., Mark Michelson wrote: branches/13/res/res_pjsip.c, lines 43-45 https://reviewboard.asterisk.org/r/4311/diff/1-2/?file=70130#file70130line43 What caused these dependencies to be added? Kevin Harwell wrote: pjsip_options uses sorcery_memory with regards to qualify/contact_status and contacts themselves are stored in the astdb (location.c) Joshua Colp wrote: I think res_sorcery_memory is fine being a dependency because it is actually required by stuff internally. res_sorcery_astdb on the other hand is only required if you are using the default of storing within astdb - it's runtime configurable to be different. George Joseph wrote: astdb is the persistent store for contacts. res_pjsip will fail to load if res_sorcery_astdb isn't loaded. Joshua Colp wrote: In a default configuration. George Joseph wrote: True but my point in asking for MODULEINFO to be updated was that it's not obvious that res_pjsip needs astdb to load in most cases. In fact, if you're setting up realtime what would even trigger you to override contact since contact isn't an object you normally define in pjsip.conf? So unless you actually looked in location.c, contact is going to need astdb. I'm fine with making it a dependency since subscription persistence does pretty much require it. Ultimately though since this is only build time dependencies and not runtime it's possible to still run into a problem. - Joshua --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4311/#review14126 --- On Jan. 9, 2015, 3:37 p.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4311/ --- (Updated Jan. 9, 2015, 3:37 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24485 https://issues.asterisk.org/jira/browse/ASTERISK-24485 Repository: Asterisk Description --- The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue by Corey Farrell with a few modifications. Removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. Diffs - branches/13/res/res_pjsip/pjsip_outbound_auth.c 430294 branches/13/res/res_pjsip/pjsip_options.c 430294 branches/13/res/res_pjsip/pjsip_global_headers.c 430294 branches/13/res/res_pjsip/pjsip_distributor.c 430294 branches/13/res/res_pjsip/pjsip_configuration.c 430294 branches/13/res/res_pjsip/location.c 430294 branches/13/res/res_pjsip/include/res_pjsip_private.h 430294 branches/13/res/res_pjsip/config_transport.c 430294 branches/13/res/res_pjsip/config_auth.c 430294 branches/13/res/res_pjsip.c 430294 branches/13/main/stasis_message_router.c 430294 Diff: https://reviewboard.asterisk.org/r/4311/diff/ Testing --- Made it so res_pjsip was the only pjsip module loaded and then issued an unload and noted it unloaded successfully (also loaded/unloaded it several times from the CLI). Also when loaded and with REF_DEBUG enabled issued a core stop gracefully and made sure there were no ref leaks for the module. Also tested unloading with other dependent pjsip modules loaded and noted that the module would not unload (as it should since dependencies are currently loaded). And then shutdown asterisk and made sure it did not crash or anything. Started asterisk with nominal and off nominal module and pjsip configurations to make sure things behaved appropriately (no crashes and such) and then attempted to, or successfully unload the res_pjsip module. Also made sure Asterisk continued to shutdown without incident. Thanks, Kevin Harwell -- _ -- 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] 4322: app_bridge: return to next dialplan priority
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4322/ --- (Updated Jan. 9, 2015, 3:45 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 430467 Bugs: ASTERISK-24637 https://issues.asterisk.org/jira/browse/ASTERISK-24637 Repository: Asterisk Description --- When app_bridge grabs a channel to put in a bridge, it should allow it to continue executing dialplan after the bridge ends. Although the current dialplan is stored as an after bridge goto on the channel, it was executing the same priority of dialplan again rather than going to the next priority. This change replaces the specific version of bridge_set_after_goto with bridge_set_after_go_on to allow the dialplan execution to naturally flow to the next priority. Diffs - /branches/13/main/features.c 430220 Diff: https://reviewboard.asterisk.org/r/4322/diff/ Testing --- Testsuite test that caught problem now passes. I also ran the bridge_baseline and other bridge tests to insure they passed. Thanks, Scott Griepentrog -- _ -- 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] 4256: testsuite: check for channel leak on failed blonde transfer
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4256/ --- (Updated Jan. 9, 2015, 3:58 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 6227 Bugs: ASTERISK-24513 https://issues.asterisk.org/jira/browse/ASTERISK-24513 Repository: testsuite Description --- This test starts an attended transfer, converts to blonde mode by hanging up the transferer, and then fails the transfer by hanging up the transferee. Then after allowing the recall attempt to complete, checks to insure that there are not leaked channels. Improvements to channel_test_condition: count the actual channels listed in core show channels output to check for leaks. Also added unittest. Diffs - /asterisk/trunk/tests/bridge/tests.yaml 6149 /asterisk/trunk/tests/bridge/atxfer_fail_blonde/test-config.yaml PRE-CREATION /asterisk/trunk/tests/bridge/atxfer_fail_blonde/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/lib/python/asterisk/channel_test_condition.py 6149 Diff: https://reviewboard.asterisk.org/r/4256/diff/ Testing --- Currently fails while ASTERISK-24513 is not yet patched. Thanks, Scott Griepentrog -- _ -- 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] 4280: sip_to_pjsip: improved ability to parse input without exception
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4280/ --- (Updated Jan. 9, 2015, 4:09 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 430469 Bugs: ASTERISK-24474 https://issues.asterisk.org/jira/browse/ASTERISK-24474 Repository: Asterisk Description --- General improvements to reliability of conversion utility: 1) track default section of input file to allow parsing an include file that doesn't specify a [section] 2) informatively handle case of assignment with no section 3) correctly handle getting sections from included files 4) assume default bind of 0.0.0.0 5) gracefully handle missing portions of registration string 6) Denote steps of operation and confirm top level conf files being read/written as a convenience Diffs - /branches/12/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py 429613 /branches/12/contrib/scripts/sip_to_pjsip/astconfigparser.py 429613 Diff: https://reviewboard.asterisk.org/r/4280/diff/ Testing --- Ran on config files from various sources to insure no exceptions occurred. Perused output to confirm appearance of converted input values. Thanks, Scott Griepentrog -- _ -- 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] 4319: testsuite: app_macro tests for channel redirect while the macro is active.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4319/ --- (Updated Jan. 9, 2015, 3:20 p.m.) Review request for Asterisk Developers. Changes --- Implemented Mark's suggestion. Bugs: ASTERISK-23850 https://issues.asterisk.org/jira/browse/ASTERISK-23850 Repository: testsuite Description --- Test channel redirect when a macro is active. 1) Redirect while in an active macro to an external dialplan location. 2) Redirect while in an active macro to an extension in the macro context. 3) Park while in an active macro and timeout to an external dialplan location. 4) Park while in an active macro and timeout to an extension in the macro context. Diffs (updated) - /asterisk/trunk/tests/apps/tests.yaml 6226 /asterisk/trunk/tests/apps/macro/tests.yaml PRE-CREATION /asterisk/trunk/tests/apps/macro/redirect_outside/test-config.yaml PRE-CREATION /asterisk/trunk/tests/apps/macro/redirect_outside/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/apps/macro/redirect_inside/test-config.yaml PRE-CREATION /asterisk/trunk/tests/apps/macro/redirect_inside/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/apps/macro/park_timeout_outside/test-config.yaml PRE-CREATION /asterisk/trunk/tests/apps/macro/park_timeout_outside/configs/ast1/res_parking.conf PRE-CREATION /asterisk/trunk/tests/apps/macro/park_timeout_outside/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/apps/macro/park_timeout_inside/test-config.yaml PRE-CREATION /asterisk/trunk/tests/apps/macro/park_timeout_inside/configs/ast1/res_parking.conf PRE-CREATION /asterisk/trunk/tests/apps/macro/park_timeout_inside/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4319/diff/ Testing --- All tests pass when the patch on review https://reviewboard.asterisk.org/r/4292/ is applied. Tests 2 and 4 fail when the patch is not applied. Thanks, rmudgett -- _ -- 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] 4325: PJSIP: Prevent slow graceful shutdown when outbound publications have not started
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4325/#review14138 --- /branches/13/res/res_pjsip_outbound_publish.c https://reviewboard.asterisk.org/r/4325/#comment24628 explicit_publish_destroy? - Joshua Colp On Jan. 9, 2015, 12:07 a.m., Mark Michelson wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4325/ --- (Updated Jan. 9, 2015, 12:07 a.m.) Review request for Asterisk Developers and Kevin Harwell. Bugs: ASTERISK-24655 https://issues.asterisk.org/jira/browse/ASTERISK-24655 Repository: Asterisk Description --- If an outbound publish is configured, and that publication never published any data, then a graceful shutdown would result in Asterisk hanging for a while before finally shutting down. The reason is that the code did not take into account the case where we never started publishing anything. The code would attempt to send a PUBLISH to stop publication, relying on the PUBLISH callback to be called so we could then destroy the PJSIP publishc structure, destroy our publication client, and signal to the unloading code that we were done. The problem is that the PUBLISH callback was never being called, presumably since pjsip_publishc_send() was failing. I modified the code to outright destroy the PJSIP publishc structure and drop its reference to our publication client if we never actually started publishing anything. This way, shutting down gracefully doesn't wait for a callback that will never occur. Also, feel free to give a better suggestion for a function name than kill_it. There are already so many variations on destroying clients in that file, I was a bit bankrupt for ideas. Diffs - /branches/13/res/res_pjsip_outbound_publish.c 430372 Diff: https://reviewboard.asterisk.org/r/4325/diff/ Testing --- Reproduced the issue originally as described in ASTERISK-24655. With the patch here, I have confirmed that the issue no longer occurs. I won't be writing a testsuite test for this, since the testsuite is not really well suited to this sort of thing. Thanks, Mark Michelson -- _ -- 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] 4311: res_pjsip: make it unloadable
On Jan. 8, 2015, 9:13 p.m., Mark Michelson wrote: branches/13/res/res_pjsip.c, lines 43-45 https://reviewboard.asterisk.org/r/4311/diff/1-2/?file=70130#file70130line43 What caused these dependencies to be added? Kevin Harwell wrote: pjsip_options uses sorcery_memory with regards to qualify/contact_status and contacts themselves are stored in the astdb (location.c) I think res_sorcery_memory is fine being a dependency because it is actually required by stuff internally. res_sorcery_astdb on the other hand is only required if you are using the default of storing within astdb - it's runtime configurable to be different. - Joshua --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4311/#review14126 --- On Jan. 8, 2015, 9:50 p.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4311/ --- (Updated Jan. 8, 2015, 9:50 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24485 https://issues.asterisk.org/jira/browse/ASTERISK-24485 Repository: Asterisk Description --- The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue by Corey Farrell with a few modifications. Removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. Diffs - branches/13/res/res_pjsip/pjsip_outbound_auth.c 430294 branches/13/res/res_pjsip/pjsip_options.c 430294 branches/13/res/res_pjsip/pjsip_global_headers.c 430294 branches/13/res/res_pjsip/pjsip_distributor.c 430294 branches/13/res/res_pjsip/pjsip_configuration.c 430294 branches/13/res/res_pjsip/location.c 430294 branches/13/res/res_pjsip/include/res_pjsip_private.h 430294 branches/13/res/res_pjsip/config_transport.c 430294 branches/13/res/res_pjsip/config_auth.c 430294 branches/13/res/res_pjsip.c 430294 branches/13/main/stasis_message_router.c 430294 Diff: https://reviewboard.asterisk.org/r/4311/diff/ Testing --- Made it so res_pjsip was the only pjsip module loaded and then issued an unload and noted it unloaded successfully (also loaded/unloaded it several times from the CLI). Also when loaded and with REF_DEBUG enabled issued a core stop gracefully and made sure there were no ref leaks for the module. Also tested unloading with other dependent pjsip modules loaded and noted that the module would not unload (as it should since dependencies are currently loaded). And then shutdown asterisk and made sure it did not crash or anything. Started asterisk with nominal and off nominal module and pjsip configurations to make sure things behaved appropriately (no crashes and such) and then attempted to, or successfully unload the res_pjsip module. Also made sure Asterisk continued to shutdown without incident. Thanks, Kevin Harwell -- _ -- 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] 4320: res_fax: Make T.38 negotiation timeout configurable and handle T.38 switch failure
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4320/ --- (Updated Jan. 9, 2015, 8:40 a.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 430415 Repository: Asterisk Description --- This change makes the T.38 negotiation timeout configurable via res_fax.conf or the FAXOPT() dialplan function. It was previously hard coded to be 5 seconds. This change also handles T.38 switch failures by aborting the fax since in the case where this can happen, both sides have agreed to switch to T.38 and Asterisk is unable to do so. Diffs - branches/11/res/res_fax.c 430372 branches/11/include/asterisk/res_fax.h 430372 branches/11/configs/res_fax.conf.sample 430372 Diff: https://reviewboard.asterisk.org/r/4320/diff/ Testing --- Manual testing and the test in review 4321. Thanks, opticron -- _ -- 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] 4321: Testsuite: Test T.38 negotiation timeout
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4321/ --- (Updated Jan. 9, 2015, 9:01 a.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 6213 Repository: testsuite Description --- This test exercises the T.38 negotiation timeout and the options used to configure it. Diffs - asterisk/trunk/tests/fax/sip/tests.yaml 6191 asterisk/trunk/tests/fax/sip/t38_negotiation_timeout/test-config.yaml PRE-CREATION asterisk/trunk/tests/fax/sip/t38_negotiation_timeout/sipp/t38timeout.xml PRE-CREATION asterisk/trunk/tests/fax/sip/t38_negotiation_timeout/configs/ast1/sip.conf PRE-CREATION asterisk/trunk/tests/fax/sip/t38_negotiation_timeout/configs/ast1/res_fax.conf PRE-CREATION asterisk/trunk/tests/fax/sip/t38_negotiation_timeout/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4321/diff/ Testing --- Verified that the test performed as expected. Thanks, opticron -- _ -- 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
