Re: [asterisk-dev] [Code Review] 4408: Testsuite: Add external bridging tests for Stasis (two channel) interactions
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4408/#review14432 --- Ship it! Excellent test descriptions! - Mark Michelson On Feb. 6, 2015, 12:05 a.m., jbigelow wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4408/ --- (Updated Feb. 6, 2015, 12:05 a.m.) Review request for Asterisk Developers and Mark Michelson. Bugs: ASTERISK-24611 https://issues.asterisk.org/jira/browse/ASTERISK-24611 Repository: testsuite Description --- This adds external bridging tests for Stasis (two channel) interactions as defined on the StasisStart/StasisEnd Test Plan (tests 2.5, 2.6, 2.7, and 2.8) at: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30279826#StasisStart/StasisEndTestplan-ExternalBridging This also renames (move to sub directory) the test 'tests/rest_api/external_interaction/ami_bridge/stasis_app/' to 'tests/rest_api/external_interaction/ami_bridge/stasis_app/non_stasis_app/'. NOTE: The files for the renamed test don't appear just because of how things work. Diffs - /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/tests.yaml 6377 /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_bridge/two_channel_same_stasis_app/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_bridge/two_channel_same_stasis_app/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_bridge/two_channel_different_stasis_app/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_bridge/two_channel_different_stasis_app/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_bridge/tests.yaml 6377 /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_app/two_channel_same_stasis_app/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_app/two_channel_same_stasis_app/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_app/two_channel_different_stasis_app/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_app/two_channel_different_stasis_app/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_app/tests.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_app/test-config.yaml 6377 /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_app/configs/ast1/extensions.conf 6377 Diff: https://reviewboard.asterisk.org/r/4408/diff/ Testing --- * Executed each test in a loop of 100 iterations with no failures. * Reviewed logs to ensure the tests were executing as expected. 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] 4411: testsuite: fix a number of tests where Asterisk does not shutdown gracefully
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4411/#review14433 --- Ship it! The only caveat here is that you may want to watch automated runs of the SIP info_dtmf test to be sure that on awful hardware the 5 second Wait() isn't too short for the test to complete. I suspect it will be okay though. On a side note, I have another test to add to my list of tests that could be rewritten to not rely on timing, though :) - Mark Michelson On Feb. 9, 2015, 5:50 p.m., Corey Farrell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4411/ --- (Updated Feb. 9, 2015, 5:50 p.m.) Review request for Asterisk Developers. Repository: testsuite Description --- * Add Hangup() to priority after Dial() where needed. This prevents auto-fallthrough from playing 10 seconds of BUSY or CONGESTION tone. * Decrease Wait(10) to Wait(5) in tests/channels/SIP/info_dtmf. * Maintain list of AGI connections where needed so they can all be agi.finish(). * Replace calls to reactor.stop() with self.stop_reactor(), remove test.start_asterisk()/test.stop_asterisk() from main(). * Delay self.stop_reactor() in tests/channels/SIP/sip_tls_call by 2 seconds. This gives the calls enough time to end and avoid shutdown timeout. Diffs - /asterisk/trunk/tests/funcs/func_srv/run-test 6377 /asterisk/trunk/tests/funcs/func_presencestate/run-test 6377 /asterisk/trunk/tests/fastagi/stream-file/run-test 6377 /asterisk/trunk/tests/fastagi/database/run-test 6377 /asterisk/trunk/tests/fastagi/control-stream-file/run-test 6377 /asterisk/trunk/tests/channels/SIP/sip_tls_call/run-test 6377 /asterisk/trunk/tests/channels/SIP/sip_srtp/run-test 6377 /asterisk/trunk/tests/channels/SIP/sip_cause/configs/ast1/extensions.conf 6377 /asterisk/trunk/tests/channels/SIP/secure_bridge_media/run-test 6377 /asterisk/trunk/tests/channels/SIP/noload_res_srtp/run-test 6377 /asterisk/trunk/tests/channels/SIP/info_dtmf/configs/ast1/extensions.conf 6377 /asterisk/trunk/tests/channels/SIP/hangupcause/configs/ast1/extensions.conf 6377 /asterisk/trunk/tests/channels/SIP/generic_ccss/configs/ast1/extensions.conf 6377 Diff: https://reviewboard.asterisk.org/r/4411/diff/ Testing --- Ran all effected tests against Asterisk 11 with REF_DEBUG. Prior to these fixes graceful shutdown of Asterisk timed out, causing reference leaks to be reported. These tests now shutdown gracefully and have no reference leaks. Thanks, Corey Farrell -- _ -- 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] 4412: res_http_websocket: websocket write timeout fails to fully disconnect underlying socket
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4412/#review14437 --- Ship it! branches/13/res/ari/ari_websockets.c https://reviewboard.asterisk.org/r/4412/#comment24957 Nitpick: this should be 0 instead of = 0. 0 is a valid fd, even though it's usually reserved for STDIN_FILENO. - Mark Michelson On Feb. 10, 2015, 6:06 p.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4412/ --- (Updated Feb. 10, 2015, 6:06 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24701 https://issues.asterisk.org/jira/browse/ASTERISK-24701 Repository: Asterisk Description --- When writing to a websocket if a timeout occurred the underlying socket did not get closed/disconnected. This patch makes sure the websocket gets disconnected on a write timeout. Also a notice is logged stating that the websocket was disconnected. Diffs - branches/13/res/res_http_websocket.c 431641 branches/13/res/ari/ari_websockets.c 431641 Diff: https://reviewboard.asterisk.org/r/4412/diff/ Testing --- I had trouble getting an actual write timeout to occur, so I just forced an error on write to make sure the underlying socket would get closed and the notice was logged. Also ran relevant testsuite and unit tests. 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] 4412: res_http_websocket: websocket write timeout fails to fully disconnect underlying socket
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4412/#review14438 --- branches/13/res/res_http_websocket.c https://reviewboard.asterisk.org/r/4412/#comment24958 Add a comment with a description of what 1011 is. I had to look it up, myself. - Joshua Colp On Feb. 10, 2015, 6:06 p.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4412/ --- (Updated Feb. 10, 2015, 6:06 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24701 https://issues.asterisk.org/jira/browse/ASTERISK-24701 Repository: Asterisk Description --- When writing to a websocket if a timeout occurred the underlying socket did not get closed/disconnected. This patch makes sure the websocket gets disconnected on a write timeout. Also a notice is logged stating that the websocket was disconnected. Diffs - branches/13/res/res_http_websocket.c 431641 branches/13/res/ari/ari_websockets.c 431641 Diff: https://reviewboard.asterisk.org/r/4412/diff/ Testing --- I had trouble getting an actual write timeout to occur, so I just forced an error on write to make sure the underlying socket would get closed and the notice was logged. Also ran relevant testsuite and unit tests. 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
[asterisk-dev] [Code Review] 4412: res_http_websocket: websocket write timeout fails to fully disconnect underlying socket
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4412/ --- Review request for Asterisk Developers. Bugs: ASTERISK-24701 https://issues.asterisk.org/jira/browse/ASTERISK-24701 Repository: Asterisk Description --- When writing to a websocket if a timeout occurred the underlying socket did not get closed/disconnected. This patch makes sure the websocket gets disconnected on a write timeout. Also a notice is logged stating that the websocket was disconnected. Diffs - branches/13/res/res_http_websocket.c 431641 branches/13/res/ari/ari_websockets.c 431641 Diff: https://reviewboard.asterisk.org/r/4412/diff/ Testing --- I had trouble getting an actual write timeout to occur, so I just forced an error on write to make sure the underlying socket would get closed and the notice was logged. Also ran relevant testsuite and unit tests. 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] 4409: res_pjsip: dtls_handler causes Asterisk to crash
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4409/#review14434 --- Ship it! Ship It! - Joshua Colp On Feb. 7, 2015, 12:25 a.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4409/ --- (Updated Feb. 7, 2015, 12:25 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24741 https://issues.asterisk.org/jira/browse/ASTERISK-24741 Repository: Asterisk Description --- There have been a couple of times where a crash occurred in the dtls_handler section of the code for res_pjsip. Unfortunately, in working this issue the problem was unable to be reproduced. After looking at the backtraces and through the code the current best guess as to why this happened might be due to a reentrance problem and the strtok function. So, the current fix is to convert the strtok function into the reentrant version of the function, strtok_r. Diffs - branches/13/res/res_pjsip/pjsip_configuration.c 431572 Diff: https://reviewboard.asterisk.org/r/4409/diff/ Testing --- Ran through the pjsip testsuite tests to make sure no crashes occurred or anything else out of the ordinary. Also while running asterisk with res_pjsip configured to use realtime issued reloads every 0.1 seconds while also executing the show endpoint command at the same interval in an attempt to potentially cause two threads to enter the dtls_handler function at the same time. No crashes occurred. 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] 4383: res_pjsip_config_wizard: Add ability to auto-create hints.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4383/#review14435 --- Ship it! branches/13/res/res_pjsip_config_wizard.c https://reviewboard.asterisk.org/r/4383/#comment24955 App and data could still be empty after this. Besides the minor finding looks good to me. - Joshua Colp On Feb. 4, 2015, 5:31 p.m., George Joseph wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4383/ --- (Updated Feb. 4, 2015, 5:31 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- Looking at the Super Awesome Company sample reminded me that creating hints is just plain gruntwork. So you can now have the pjsip conifg wizard auto-create them for you. Specifying 'hint_exten' in the wizard will create 'exten = hint_exten,hint/PJSIP/wizard_id' in whatever is specified for 'hint_context'. Specifying 'hint_application' in the wizard will create 'exten = hint_exten,1,hint_application' in whatever is specified for 'hint_context'. The default for 'hint_context' is the endpoint's context. There's no default for 'hint_application'. If not specified, no app is added. There's no default for 'hint_exten'. If not specified, neither the hint itself nor the application will be created. Some may think this is the slippery slope to users.conf but hints are a basic necessity for phones unlike voicemail, manager, etc that users.conf creates. Diffs - branches/13/res/res_pjsip_config_wizard.c 431571 branches/13/configs/samples/pjsip_wizard.conf.sample 431571 Diff: https://reviewboard.asterisk.org/r/4383/diff/ Testing --- Existing config_wizard testsuite tests pass. Additional testsuite tests in the works. Thanks, George Joseph -- _ -- 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] 4399: HTTP: Stop accepting requests on final system shutdown.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4399/#review14439 --- Ship it! Ship It! - Mark Michelson On Feb. 5, 2015, 9:58 p.m., rmudgett wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4399/ --- (Updated Feb. 5, 2015, 9:58 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24752 https://issues.asterisk.org/jira/browse/ASTERISK-24752 Repository: Asterisk Description --- There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. Diffs - /branches/13/res/res_pjsip_pubsub.c 431574 /branches/13/res/res_pjsip/pjsip_options.c 431574 /branches/13/main/http.c 431574 /branches/13/main/channel.c 431574 /branches/13/main/bridge.c 431574 /branches/13/main/asterisk.c 431574 /branches/13/include/asterisk/channel.h 431574 /branches/13/include/asterisk.h 431574 /branches/13/channels/chan_sip.c 431574 /branches/13/apps/app_confbridge.c 431574 Diff: https://reviewboard.asterisk.org/r/4399/diff/ Testing --- Extended the final shutdown phase sleep so I could send a HTTP request while in the final shutdown phase. The HTTP request was not refused while the shutdown request was pending and refused after the final shutdown phase was reached. 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] 4383: res_pjsip_config_wizard: Add ability to auto-create hints.
On Feb. 10, 2015, 12:50 p.m., Joshua Colp wrote: branches/13/res/res_pjsip_config_wizard.c, lines 427-430 https://reviewboard.asterisk.org/r/4383/diff/3/?file=71271#file71271line427 App and data could still be empty after this. Not to mention that data is not NULL checked. - rmudgett --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4383/#review14435 --- On Feb. 4, 2015, 11:31 a.m., George Joseph wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4383/ --- (Updated Feb. 4, 2015, 11:31 a.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- Looking at the Super Awesome Company sample reminded me that creating hints is just plain gruntwork. So you can now have the pjsip conifg wizard auto-create them for you. Specifying 'hint_exten' in the wizard will create 'exten = hint_exten,hint/PJSIP/wizard_id' in whatever is specified for 'hint_context'. Specifying 'hint_application' in the wizard will create 'exten = hint_exten,1,hint_application' in whatever is specified for 'hint_context'. The default for 'hint_context' is the endpoint's context. There's no default for 'hint_application'. If not specified, no app is added. There's no default for 'hint_exten'. If not specified, neither the hint itself nor the application will be created. Some may think this is the slippery slope to users.conf but hints are a basic necessity for phones unlike voicemail, manager, etc that users.conf creates. Diffs - branches/13/res/res_pjsip_config_wizard.c 431571 branches/13/configs/samples/pjsip_wizard.conf.sample 431571 Diff: https://reviewboard.asterisk.org/r/4383/diff/ Testing --- Existing config_wizard testsuite tests pass. Additional testsuite tests in the works. Thanks, George Joseph -- _ -- 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] 4383: res_pjsip_config_wizard: Add ability to auto-create hints.
On Feb. 10, 2015, 11:50 a.m., Joshua Colp wrote: branches/13/res/res_pjsip_config_wizard.c, lines 427-430 https://reviewboard.asterisk.org/r/4383/diff/3/?file=71271#file71271line427 App and data could still be empty after this. rmudgett wrote: Not to mention that data is not NULL checked. I'll add the checks before I commit. - George --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4383/#review14435 --- On Feb. 4, 2015, 10:31 a.m., George Joseph wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4383/ --- (Updated Feb. 4, 2015, 10:31 a.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- Looking at the Super Awesome Company sample reminded me that creating hints is just plain gruntwork. So you can now have the pjsip conifg wizard auto-create them for you. Specifying 'hint_exten' in the wizard will create 'exten = hint_exten,hint/PJSIP/wizard_id' in whatever is specified for 'hint_context'. Specifying 'hint_application' in the wizard will create 'exten = hint_exten,1,hint_application' in whatever is specified for 'hint_context'. The default for 'hint_context' is the endpoint's context. There's no default for 'hint_application'. If not specified, no app is added. There's no default for 'hint_exten'. If not specified, neither the hint itself nor the application will be created. Some may think this is the slippery slope to users.conf but hints are a basic necessity for phones unlike voicemail, manager, etc that users.conf creates. Diffs - branches/13/res/res_pjsip_config_wizard.c 431571 branches/13/configs/samples/pjsip_wizard.conf.sample 431571 Diff: https://reviewboard.asterisk.org/r/4383/diff/ Testing --- Existing config_wizard testsuite tests pass. Additional testsuite tests in the works. Thanks, George Joseph -- _ -- 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
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I will not be in the office till Feb 24, but I will be checking my email periodically. -- _ -- 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] 4413: Testsuite: Simulate phones and control from YAML.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4413/ --- Review request for Asterisk Developers. Repository: testsuite Description --- Pluggable modules to place, receive, and transfer (blind/attended) calls to simulate phones using PJSUA and YAML configuration. Calls are placed and/or transferred using the new pluggable action module. This should allow many currrent and future tests to easily send/receive calls to/from Asterisk along with transferring calls within YAML configuration. See attached file for YAML demo. Diffs - /asterisk/trunk/lib/python/asterisk/pluggable_modules.py 6379 /asterisk/trunk/lib/python/asterisk/pjsua_mod.py 6379 /asterisk/trunk/lib/python/asterisk/phones.py PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4413/diff/ Testing --- * Tested placing calls, receiving calls, transfering via blind attended. * Pylint score of 9.40/10 for phones.py * See attached test-config.yaml for a demonstration. File Attachments Demonstration https://reviewboard.asterisk.org/media/uploaded/files/2015/02/11/659ab31f-8401-4f24-be5e-da1db0be3156__test-config.yaml 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] 4413: Testsuite: Simulate phones and control from YAML.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4413/ --- (Updated Feb. 11, 2015, 12:32 a.m.) Review request for Asterisk Developers. Repository: testsuite Description --- Pluggable modules to place, receive, and transfer (blind/attended) calls to simulate phones using PJSUA and YAML configuration. Calls are placed and/or transferred using the new pluggable action module. This should allow many currrent and future tests to easily send/receive calls to/from Asterisk along with transferring calls within YAML configuration. See attached file for YAML demo. Diffs - /asterisk/trunk/lib/python/asterisk/pluggable_modules.py 6379 /asterisk/trunk/lib/python/asterisk/pjsua_mod.py 6379 /asterisk/trunk/lib/python/asterisk/phones.py PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4413/diff/ Testing --- * Tested placing calls, receiving calls, transfering via blind attended. * Pylint score of 9.40/10 for phones.py * See attached test-config.yaml for a demonstration. File Attachments Demonstration https://reviewboard.asterisk.org/media/uploaded/files/2015/02/11/659ab31f-8401-4f24-be5e-da1db0be3156__test-config.yaml 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] 4383: res_pjsip_config_wizard: Add ability to auto-create hints.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4383/ --- (Updated Feb. 10, 2015, 5:16 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 431643 Repository: Asterisk Description --- Looking at the Super Awesome Company sample reminded me that creating hints is just plain gruntwork. So you can now have the pjsip conifg wizard auto-create them for you. Specifying 'hint_exten' in the wizard will create 'exten = hint_exten,hint/PJSIP/wizard_id' in whatever is specified for 'hint_context'. Specifying 'hint_application' in the wizard will create 'exten = hint_exten,1,hint_application' in whatever is specified for 'hint_context'. The default for 'hint_context' is the endpoint's context. There's no default for 'hint_application'. If not specified, no app is added. There's no default for 'hint_exten'. If not specified, neither the hint itself nor the application will be created. Some may think this is the slippery slope to users.conf but hints are a basic necessity for phones unlike voicemail, manager, etc that users.conf creates. Diffs - branches/13/res/res_pjsip_config_wizard.c 431571 branches/13/configs/samples/pjsip_wizard.conf.sample 431571 Diff: https://reviewboard.asterisk.org/r/4383/diff/ Testing --- Existing config_wizard testsuite tests pass. Additional testsuite tests in the works. Thanks, George Joseph -- _ -- 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
