Re: [asterisk-dev] [Code Review] 3476: Memory Pre- and Post-Test Condition
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3476/#review11739 --- ./asterisk/trunk/lib/python/asterisk/memory_test_condition.py https://reviewboard.asterisk.org/r/3476/#comment21532 Why math.fabs() and not just abs() ? Aren't you dealing with integers here? I don't share Marks fear that decreasing memory usage would create false positives. If that were to be an issue, we can always go back and alter the test. ./asterisk/trunk/lib/python/asterisk/memory_test_condition.py https://reviewboard.asterisk.org/r/3476/#comment21533 E128: You want to line these up past the parenthesis: x = (abc, def) - wdoekes On April 24, 2014, 7:02 p.m., Benjamin Keith Ford wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3476/ --- (Updated April 24, 2014, 7:02 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-18429 https://issues.asterisk.org/jira/browse/ASTERISK-18429 Repository: testsuite Description --- This testcondition can be enabled for any test using the keyword 'memory' under testconditions. The purpose of this testcondition is to check the memory allocated before and after the test, and make sure they are within a certain range. If the test wants to look at something specific (such as channel.c), then each allocation that you want to look at can also be specified in under 'allocations'. If both the global memory and individual allocations are to be checked by the test, that option can be enabled by setting 'both' to value True. Diffs - ./asterisk/trunk/test-config.yaml 4944 ./asterisk/trunk/lib/python/asterisk/test_conditions.py 4944 ./asterisk/trunk/lib/python/asterisk/test_case.py 4944 ./asterisk/trunk/lib/python/asterisk/memory_test_condition.py PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3476/diff/ Testing --- Thanks, Benjamin Keith Ford -- _ -- 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] encoding issues in Asterisk 11.9.0 Now Available
On 23/04/14 18:52, Asterisk Development Team wrote: --===4365525224653466459== Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 8bit ... * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted minus signs (Reported by Jeremy Lainé) ... * ASTERISK-19499 - ConfBridge MOH is not working for transferee after attended transfer (Reported by Timo Teräs) ... Could you update the `charset` param to utf-8 the next time? Thanks! /w -- _ -- 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] 3477: Japanese language patch for app_voicemail.c and say.c
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3477/ --- Review request for Asterisk Developers. Repository: Asterisk Description --- Created new review request with proper diffs. Japanese language support patch for sound files in app_voicemail.c and say.c, depends on entire sound file package currently available in release candidate format as per https://issues.asterisk.org/jira/browse/ASTERISK-23324 https://www.dropbox.com/s/axu6gfnf9fh40hz/asterisk-core-sounds-ja-wav-and-patch.tgz Word order and plurals, dates, counts in Japanese are all significantly different than English, hence the need for this patch. Tested working with Asterisk 12. Here is the installation procedure from the README contained in the RC archive: -- Install Asterisk Sound Files: mkdir /var/lib/asterisk/sounds/ja cd /var/lib/asterisk/sounds/ja wget http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-ja-gsm-current.tar.gz tar xvfz asterisk-core-sounds-ja-gsm-current.tar.gz rm -f asterisk-core-sounds-ja-gsm-current.tar.gz chown -R asterisk.asterisk /var/lib/asterisk/sounds/ja How To Change Default SIP Channel Language to Japanese Using Asterisk (vanilla): vi /etc/asterisk/sip.conf Edit language variable to: language = ja Using FreePBX: On the FreePBX menu - Settings - Asterisk SIP Settings - Advanced General Settings section - Language field Set Language field to : ja Install Japanese Patch: Download Asterisk 12 source Change directory to Asterisk 12 source folder Download the patches for say.c and app_voicemail.c patch -p0 say.c.20140226.jp.patch patch -p0 app_voicemail.c.20140226.jp.patch Compile Asterisk 12 source as usual -- I'm happy to answer questions about the code. Diffs - /trunk/main/say.c 413007 /trunk/apps/app_voicemail.c 413007 Diff: https://reviewboard.asterisk.org/r/3477/diff/ Testing --- Thanks, Kevin McCoy -- _ -- 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] 3443: Japanese language patch for app_voicemail.c and say.c, compatible with newly submitted Japanese sound files
On April 19, 2014, 4:07 p.m., Matt Jordan wrote: It looks like these patches were merely attached to the review board posting, as opposed to being uploaded using post-review or uploaded directly as a diff. That makes it very hard to review. Please review the instructions for using Review Board (https://wiki.asterisk.org/wiki/display/AST/Review+Board+Usage) and post the diffs so that they show up as code to be reviewed. Thanks for the info, I had to recreate a review request since I was getting error 500 when trying to add to this one. Also not sure if you were aware, but the documentation on that link is not current with the latest commands in RBTools; post-review is now rbt post. - Kevin --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3443/#review11705 --- On April 14, 2014, 5:34 a.m., Kevin McCoy wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3443/ --- (Updated April 14, 2014, 5:34 a.m.) Review request for Asterisk Developers. Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-23324 None Description --- Japanese language support patch for sound files in app_voicemail.c and say.c, depends on entire sound file package currently available in release candidate format as per https://issues.asterisk.org/jira/browse/ASTERISK-23324 https://www.dropbox.com/s/axu6gfnf9fh40hz/asterisk-core-sounds-ja-wav-and-patch.tgz Word order and plurals, dates, counts in Japanese are all significantly different than English, hence the need for this patch. Tested working with Asterisk 12. Here is the installation procedure from the README contained in the RC archive: -- Install Asterisk Sound Files: mkdir /var/lib/asterisk/sounds/ja cd /var/lib/asterisk/sounds/ja wget http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-ja-gsm-current.tar.gz tar xvfz asterisk-core-sounds-ja-gsm-current.tar.gz rm -f asterisk-core-sounds-ja-gsm-current.tar.gz chown -R asterisk.asterisk /var/lib/asterisk/sounds/ja How To Change Default SIP Channel Language to Japanese Using Asterisk (vanilla): vi /etc/asterisk/sip.conf Edit language variable to: language = ja Using FreePBX: On the FreePBX menu - Settings - Asterisk SIP Settings - Advanced General Settings section - Language field Set Language field to : ja Install Japanese Patch: Download Asterisk 12 source Change directory to Asterisk 12 source folder Download the patches for say.c and app_voicemail.c patch -p0 say.c.20140226.jp.patch patch -p0 app_voicemail.c.20140226.jp.patch Compile Asterisk 12 source as usual -- I'm happy to answer questions about the code. Diffs - Diff: https://reviewboard.asterisk.org/r/3443/diff/ Testing --- File Attachments app_voicemail.c.20140226.jp.patch https://reviewboard.asterisk.org/media/uploaded/files/2014/04/14/cfe35d8c-066d-4b69-9f52-8e104f8b3770__app_voicemail.c.20140226.jp.patch say.c.20140226.jp.patch https://reviewboard.asterisk.org/media/uploaded/files/2014/04/14/2f6dd68b-7549-48d8-8750-e043202f41f5__say.c.20140226.jp.patch Thanks, Kevin McCoy -- _ -- 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] 3443: Japanese language patch for app_voicemail.c and say.c, compatible with newly submitted Japanese sound files
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3443/ --- (Updated April 25, 2014, 10:17 a.m.) Status -- This change has been discarded. Review request for Asterisk Developers. Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-23324 None Description --- Japanese language support patch for sound files in app_voicemail.c and say.c, depends on entire sound file package currently available in release candidate format as per https://issues.asterisk.org/jira/browse/ASTERISK-23324 https://www.dropbox.com/s/axu6gfnf9fh40hz/asterisk-core-sounds-ja-wav-and-patch.tgz Word order and plurals, dates, counts in Japanese are all significantly different than English, hence the need for this patch. Tested working with Asterisk 12. Here is the installation procedure from the README contained in the RC archive: -- Install Asterisk Sound Files: mkdir /var/lib/asterisk/sounds/ja cd /var/lib/asterisk/sounds/ja wget http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-ja-gsm-current.tar.gz tar xvfz asterisk-core-sounds-ja-gsm-current.tar.gz rm -f asterisk-core-sounds-ja-gsm-current.tar.gz chown -R asterisk.asterisk /var/lib/asterisk/sounds/ja How To Change Default SIP Channel Language to Japanese Using Asterisk (vanilla): vi /etc/asterisk/sip.conf Edit language variable to: language = ja Using FreePBX: On the FreePBX menu - Settings - Asterisk SIP Settings - Advanced General Settings section - Language field Set Language field to : ja Install Japanese Patch: Download Asterisk 12 source Change directory to Asterisk 12 source folder Download the patches for say.c and app_voicemail.c patch -p0 say.c.20140226.jp.patch patch -p0 app_voicemail.c.20140226.jp.patch Compile Asterisk 12 source as usual -- I'm happy to answer questions about the code. Diffs - Diff: https://reviewboard.asterisk.org/r/3443/diff/ Testing --- File Attachments app_voicemail.c.20140226.jp.patch https://reviewboard.asterisk.org/media/uploaded/files/2014/04/14/cfe35d8c-066d-4b69-9f52-8e104f8b3770__app_voicemail.c.20140226.jp.patch say.c.20140226.jp.patch https://reviewboard.asterisk.org/media/uploaded/files/2014/04/14/2f6dd68b-7549-48d8-8750-e043202f41f5__say.c.20140226.jp.patch Thanks, Kevin McCoy -- _ -- 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] 3478: chan_pjsip: Add call pickup support.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3478/ --- Review request for Asterisk Developers. Repository: Asterisk Description --- While configuration exists to place PJSIP channels into pickup and call groups the functionality to actually perform a call pickup does not exist. This change adds it. Diffs - /branches/12/res/res_pjsip_session.c 413007 /branches/12/channels/chan_pjsip.c 413007 Diff: https://reviewboard.asterisk.org/r/3478/diff/ Testing --- Ran test and confirmed failed on normal 12. Applied change. Re-ran test and confirmed fixed. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] [Code Review] 3479: chan_pjsip: Call pickup test.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3479/ --- Review request for Asterisk Developers. Repository: testsuite Description --- This is a modified version of the normal call pickup test which uses chan_pjsip instead of chan_sip to test call pickup functionality. Diffs - /asterisk/trunk/tests/channels/pjsip/call_pickup/test-config.yaml PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast2/pjsip.conf PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast2/extensions.conf PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast1/pjsip.conf PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast1/features.conf PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3479/diff/ Testing --- I tested the test by running the test. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Add new option to Queue function
Hi all, I'm using Queue function of Asterisk to arrange calls which is coming to my agents. I want to customize the way asterisk arrange coming call, in other word, is it possible to create a new option instead of using the existing: RINGALL, ROUNDROBIN,... . For example: The incoming call should come to the argent who has the most waiting time (idle time). I find out that the algorithm of each option of Queue is defined in app_queue.c in the source code but I don't know how to change, how to add the waiting time as a new option to sort by. This question is quite related to the development of asterisk, so please help if you have any idea or experience on that. Thank you very much. --- NGUYỄN HOÀNG SƠN M-Commerce Center VASC Software and Media Company - VNPT Addr: No.97 Nguyen Chi Thanh Street, Dong Da District, Hanoi, Vietnam Cell phone: +84 912998101 Skype: hoangsonk49 E-mail: nh...@vasc.com.vn -- _ -- 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] encoding issues in Asterisk 11.9.0 Now Available
On Fri, Apr 25, 2014 at 4:32 AM, Walter Doekes walter+asterisk-...@osso.nl wrote: On 23/04/14 18:52, Asterisk Development Team wrote: --===4365525224653466459== Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 8bit ... * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted minus signs (Reported by Jeremy Lainé) ... * ASTERISK-19499 - ConfBridge MOH is not working for transferee after attended transfer (Reported by Timo Teräs) ... Could you update the `charset` param to utf-8 the next time? Thanks! Sure - sorry about that! -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] 3478: chan_pjsip: Add call pickup support.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3478/#review11741 --- Ship it! /branches/12/res/res_pjsip_session.c https://reviewboard.asterisk.org/r/3478/#comment21534 Blob. - Matt Jordan On April 25, 2014, 8:06 a.m., Joshua Colp wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3478/ --- (Updated April 25, 2014, 8:06 a.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- While configuration exists to place PJSIP channels into pickup and call groups the functionality to actually perform a call pickup does not exist. This change adds it. Diffs - /branches/12/res/res_pjsip_session.c 413007 /branches/12/channels/chan_pjsip.c 413007 Diff: https://reviewboard.asterisk.org/r/3478/diff/ Testing --- Ran test and confirmed failed on normal 12. Applied change. Re-ran test and confirmed fixed. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Add new option to Queue function
Just something I know which may restrict what can be done. Avaya have many patents for call distribution. This includes call distribution to agents who have spent the least amount of time on the phone and taken the lowest number of calls. On 25 Apr 2014 15:00, Nguyen Hoang Son nh...@vasc.com.vn wrote: Hi all, I'm using Queue function of Asterisk to arrange calls which is coming to my agents. I want to customize the way asterisk arrange coming call, in other word, is it possible to create a new option instead of using the existing: RINGALL, ROUNDROBIN,... . For example: The incoming call should come to the argent who has the most waiting time (idle time). I find out that the algorithm of each option of Queue is defined in app_queue.c in the source code but I don't know how to change, how to add the waiting time as a new option to sort by. This question is quite related to the development of asterisk, so please help if you have any idea or experience on that. Thank you very much. --- *NGUYỄN HOÀNG SƠN* M-Commerce Center VASC Software and Media Company - VNPT Addr: No.97 Nguyen Chi Thanh Street, Dong Da District, Hanoi, Vietnam Cell phone: +84 912998101 Skype: hoangsonk49 E-mail: nh...@vasc.com.vn -- _ -- 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 -- _ -- 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] 3479: chan_pjsip: Call pickup test.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3479/#review11742 --- /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test https://reviewboard.asterisk.org/r/3479/#comment21536 2014 /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test https://reviewboard.asterisk.org/r/3479/#comment21535 Are you sure you're Jonathan Rose? /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test https://reviewboard.asterisk.org/r/3479/#comment21543 These are always used as regular expressions. Why not just compile them here and use them as such everywhere else? /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test https://reviewboard.asterisk.org/r/3479/#comment21537 Since this is using PJSIP, there's no need to support previous versions of Asterisk. Just the bridging model for 12 is sufficient. /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test https://reviewboard.asterisk.org/r/3479/#comment21540 And just use 12 here as well /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test https://reviewboard.asterisk.org/r/3479/#comment21541 No spaces between parameters and their values: channel=Local/test_out@pickuptest /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test https://reviewboard.asterisk.org/r/3479/#comment21538 Just use the Asterisk 12 logic /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test https://reviewboard.asterisk.org/r/3479/#comment21539 PEP8 Guidelines: no spaces between equals in parameters passed to a function. You may want to pass this through pylint to catch anything else as well. /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test https://reviewboard.asterisk.org/r/3479/#comment21542 Why is the Local channel shouting at me? :-) - Matt Jordan On April 25, 2014, 8:05 a.m., Joshua Colp wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3479/ --- (Updated April 25, 2014, 8:05 a.m.) Review request for Asterisk Developers. Repository: testsuite Description --- This is a modified version of the normal call pickup test which uses chan_pjsip instead of chan_sip to test call pickup functionality. Diffs - /asterisk/trunk/tests/channels/pjsip/call_pickup/test-config.yaml PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast2/pjsip.conf PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast2/extensions.conf PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast1/pjsip.conf PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast1/features.conf PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3479/diff/ Testing --- I tested the test by running the test. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3449: Testsuite: PJSIPQualify AMI Action Test
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3449/ --- (Updated April 25, 2014, 9:27 a.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 4984 Bugs: ASTERISK-23534 https://issues.asterisk.org/jira/browse/ASTERISK-23534 Repository: testsuite Description --- This test registers an endpoint with Asterisk using a SIPp scenario, receives a 200 OK, sends the PJSIPQualify Action to Asterisk, the endpoint receives an options request, then finally returns a 200 OK. Diffs - ./asterisk/trunk/tests/channels/pjsip/ami/tests.yaml 4957 ./asterisk/trunk/tests/channels/pjsip/ami/pjsip_qualify/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/channels/pjsip/ami/pjsip_qualify/sipp/options.xml PRE-CREATION ./asterisk/trunk/tests/channels/pjsip/ami/pjsip_qualify/configs/ast1/pjsip.conf PRE-CREATION ./asterisk/trunk/tests/channels/pjsip/ami/pjsip_qualify/AMISendTest.py PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3449/diff/ Testing --- Thanks, Scott Emidy -- _ -- 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] 3457: DISA Test - Invalid Extension
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3457/ --- (Updated April 25, 2014, 9:32 a.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 4985 Bugs: ASTERISK-23526 https://issues.asterisk.org/jira/browse/ASTERISK-23526 Repository: testsuite Description --- This test has a local channel enter the DISA application, enters in an extension that does not exist, and sent to the 'i' extension to be hung up. Diffs - ./asterisk/trunk/tests/apps/disa/nominal/tests.yaml 4944 ./asterisk/trunk/tests/apps/disa/nominal/invalid_exten/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/apps/disa/nominal/invalid_exten/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3457/diff/ Testing --- Thanks, Benjamin Keith Ford -- _ -- 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] 3476: Memory Pre- and Post-Test Condition
On April 24, 2014, 10:46 p.m., Mark Michelson wrote: I suggest writing a sample yaml file that illustrates how this is intended to be used and explains what the default values are for the various configuration options. I'll start working on that. In the mean time, I'm going to go ahead and upload all the revisions just in case any more findings pop up. On April 24, 2014, 10:46 p.m., Mark Michelson wrote: ./asterisk/trunk/lib/python/asterisk/test_conditions.py, lines 122-123 https://reviewboard.asterisk.org/r/3476/diff/1/?file=57814#file57814line122 Any particular reason you switched this away from raising an exception? This was changed by Matt when he found that post conditions were not registering properly. I'm assuming this change was made because it's not a ValueError if a pre condition isn't found at all, since a ValueError should be raised if the right type is found. If nothing is found at all, than a log error or something similar would be more appropriate. - Benjamin --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3476/#review11734 --- On April 25, 2014, 2:46 p.m., Benjamin Keith Ford wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3476/ --- (Updated April 25, 2014, 2:46 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-18429 https://issues.asterisk.org/jira/browse/ASTERISK-18429 Repository: testsuite Description --- This testcondition can be enabled for any test using the keyword 'memory' under testconditions. The purpose of this testcondition is to check the memory allocated before and after the test, and make sure they are within a certain range. If the test wants to look at something specific (such as channel.c), then each allocation that you want to look at can also be specified in under 'allocations'. If both the global memory and individual allocations are to be checked by the test, that option can be enabled by setting 'both' to value True. Diffs - ./asterisk/trunk/test-config.yaml 4944 ./asterisk/trunk/lib/python/asterisk/test_conditions.py 4944 ./asterisk/trunk/lib/python/asterisk/test_case.py 4944 ./asterisk/trunk/lib/python/asterisk/memory_test_condition.py PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3476/diff/ Testing --- Thanks, Benjamin Keith Ford -- _ -- 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] 3476: Memory Pre- and Post-Test Condition
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3476/ --- (Updated April 25, 2014, 2:46 p.m.) Review request for Asterisk Developers. Changes --- - Updated comments to be more appropriate - Made some efficiency changes - Fixed logic in post check - Lined up code appropriately Bugs: ASTERISK-18429 https://issues.asterisk.org/jira/browse/ASTERISK-18429 Repository: testsuite Description --- This testcondition can be enabled for any test using the keyword 'memory' under testconditions. The purpose of this testcondition is to check the memory allocated before and after the test, and make sure they are within a certain range. If the test wants to look at something specific (such as channel.c), then each allocation that you want to look at can also be specified in under 'allocations'. If both the global memory and individual allocations are to be checked by the test, that option can be enabled by setting 'both' to value True. Diffs (updated) - ./asterisk/trunk/test-config.yaml 4944 ./asterisk/trunk/lib/python/asterisk/test_conditions.py 4944 ./asterisk/trunk/lib/python/asterisk/test_case.py 4944 ./asterisk/trunk/lib/python/asterisk/memory_test_condition.py PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3476/diff/ Testing --- Thanks, Benjamin Keith Ford -- _ -- 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] 3420: Testsuite: Call Files' Max Retries
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3420/#review11743 --- ./asterisk/trunk/lib/python/asterisk/pluggable_modules.py https://reviewboard.asterisk.org/r/3420/#comment21550 Is each configuration meant to run multiple call file scenarios at a time? Based on the rest of the code it all looks setup to only do one. If that's the case the for loop is not needed. ./asterisk/trunk/lib/python/asterisk/pluggable_modules.py https://reviewboard.asterisk.org/r/3420/#comment21544 This still only sets a single call file params. If multiple call-file-params are configured then this will only set the last value. If it is meant to only run a single test then remove the loop otherwise you'll need to find a way to store multiple configured call file params and have them looked back up on the user event. ./asterisk/trunk/lib/python/asterisk/pluggable_modules.py https://reviewboard.asterisk.org/r/3420/#comment21545 You only need to register the observer once. ./asterisk/trunk/lib/python/asterisk/pluggable_modules.py https://reviewboard.asterisk.org/r/3420/#comment21546 Again, this works because you are currently running one test, but if multiple configurations are specified then this will only use the index of the last test. ./asterisk/trunk/tests/pbx/call_file_retries_alwaysdelete/retries_alwaysdelete.py https://reviewboard.asterisk.org/r/3420/#comment21548 Is there a reason the handler(s) have to be called later? - Kevin Harwell On April 24, 2014, 5:13 p.m., Scott Emidy wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3420/ --- (Updated April 24, 2014, 5:13 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-23218 https://issues.asterisk.org/jira/browse/ASTERISK-23218 Repository: testsuite Description --- These tests involved checking that call files max retries are functioning as planned through four tests: 1) The first test (call_file_retries_fail) required that the call file originates a local channel to a dialplan extension that will always fail, and checks to make sure that it ran through all of its max retries. 2) The second test (call_file_retries_success) involves a call file that originates a local channel that will fail once, but then is answered before it hits its max retries. 3) The third test (call_file_retries_alwaysdelete) consists of checking whether or not the call file was deleted from the [astspooldir]'s outgoing folder when the alwaysdelete option is set to 'no'. 4) The fourth and final test (call_file_retries_archive) consists of checking whether or not the call file was placed in [astspooldir]'s outgoing_done folder when archive is set to 'yes'. Diffs - ./asterisk/trunk/tests/pbx/tests.yaml 4983 ./asterisk/trunk/tests/pbx/call_file_retries_success/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_success/retries_success.py PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_success/configs/ast1/extensions.conf PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_fail/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_fail/retries_fail.py PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_fail/configs/ast1/extensions.conf PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_archive/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_archive/retries_archive.py PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_archive/configs/ast1/extensions.conf PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_alwaysdelete/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_alwaysdelete/retries_alwaysdelete.py PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_alwaysdelete/configs/ast1/extensions.conf PRE-CREATION ./asterisk/trunk/lib/python/asterisk/pluggable_modules.py 4983 Diff: https://reviewboard.asterisk.org/r/3420/diff/ Testing --- Thanks, Scott Emidy -- _ -- 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] 3476: Memory Pre- and Post-Test Condition
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3476/ --- (Updated April 25, 2014, 3:41 p.m.) Review request for Asterisk Developers. Changes --- - Added a yaml sample that explains the usage of this test condition and an example Bugs: ASTERISK-18429 https://issues.asterisk.org/jira/browse/ASTERISK-18429 Repository: testsuite Description --- This testcondition can be enabled for any test using the keyword 'memory' under testconditions. The purpose of this testcondition is to check the memory allocated before and after the test, and make sure they are within a certain range. If the test wants to look at something specific (such as channel.c), then each allocation that you want to look at can also be specified in under 'allocations'. If both the global memory and individual allocations are to be checked by the test, that option can be enabled by setting 'both' to value True. Diffs (updated) - ./asterisk/trunk/test-config.yaml 4944 ./asterisk/trunk/sample-yaml/memorytestcondition-config.yaml.sample PRE-CREATION ./asterisk/trunk/lib/python/asterisk/test_conditions.py 4944 ./asterisk/trunk/lib/python/asterisk/test_case.py 4944 ./asterisk/trunk/lib/python/asterisk/memory_test_condition.py PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3476/diff/ Testing --- Thanks, Benjamin Keith Ford -- _ -- 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] 3480: chan_pjsip: Implement get_pvt_uniqueid channel technology callback.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3480/ --- Review request for Asterisk Developers. Repository: Asterisk Description --- This change implements the get_pvt_uniqueid channel technology callback in chan_pjsip which returns the call-id of the underlying dialog in use. Diffs - /branches/12/channels/chan_pjsip.c 413007 Diff: https://reviewboard.asterisk.org/r/3480/diff/ Testing --- Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3446: Parking: Add 'AnnounceChannel' to Park manager action. Change some announcement behavior for Park manager action.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3446/ --- (Updated April 25, 2014, 11:54 a.m.) Review request for Asterisk Developers, Matt Jordan and rmudgett. Changes --- Address findings Bugs: ASTERISK-23397 https://issues.asterisk.org/jira/browse/ASTERISK-23397 Repository: Asterisk Description --- r334840 removed announcements from Park manager actions back in 2011 from all of the actively supported Asterisk versions. Asterisk 12 has provided an opportunity to bring this functionality back. TimeoutChannel will now receive announcements under the strict condition that it is in a one to one bridge with Channel (the channel being parked) at the time the Park action was invoked. In this case, TimeoutChannel will be treated more or less entirely as the channel responsible for parking the call instead of just as a return point for when the call times out. Parking behavior in cases where TimeoutChannel isn't directly bridged with Channel remains mostly unchanged. The channel being parked will no longer receive announcements, but it will still be treated as having more or less self-parked. Timeout Channel will still work just as a comeback override at that point (Will be used to dial when the call times out if it's specified). AnnounceChannel field has been added to the Park action. If the AnnounceChannel field is specified and maps to an active channel, a parking announcement listener stasis subscription will be applied to that channel. When Channel is parked, that listener will trip and apply the announcement bridge feature to the AnnounceChannel. Provided that AnnounceChannel is in some kind of bridge that can use features at that point (tested with two party bridges and holding bridges), the AnnounceChannel will receive the parking announcement while staying on the bridge. If AnnounceChannel and TimeoutChannel are the same channel and that channel is bridged with Channel, a safeguard is in place to make sure multiple announcements aren't queued. In that case, AnnounceChannel is just ignored. Diffs (updated) - /branches/12/res/parking/res_parking.h 412989 /branches/12/res/parking/parking_manager.c 412989 /branches/12/res/parking/parking_bridge_features.c 412989 /branches/12/CHANGES 412989 Diff: https://reviewboard.asterisk.org/r/3446/diff/ Testing --- Tested Parking with the park action using different parking lot and timeout settings under the following scenarios: ___ Channel: SIP channel in a holding bridge TimeoutChannel: SIP channel in another holding bridge AnnounceChannel: same as TimeoutChannel Results: Timeout Channel received announcements, remained in holding bridge, and was set as the comeback dial channel. Channel gets dialed upon timeout. --- Channel: SIP channel talking to TimeoutChannel TimeoutChannel: SIP channel talking to Channel AnnounceChannel: both unspecified and the same as TimeoutChannel Results: TimeoutChannel received announcements and then hung up... treated as the Parker of the call. Gets dialed after timeout. --- Channel: Local channel in a Holding Bridge TimeoutChannel: SIP channel talking to another, unrelated SIP channel AnnounceChannel: Same as TimeoutChannel Results: TimeoutChannel receives announcements, acts as comeback dial channel. --- Channel: Local channel in a Holding Bridge TimeoutChannel: SIP channel talking to another, unrelated SIP channel AnnounceChannel: Unspecified Results: SIP channel acts as comeback dial channel, but does not receive announcements Thanks, Jonathan Rose -- _ -- 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] 3446: Parking: Add 'AnnounceChannel' to Park manager action. Change some announcement behavior for Park manager action.
On April 24, 2014, 5:21 p.m., rmudgett wrote: /branches/12/res/parking/parking_manager.c, lines 461-463 https://reviewboard.asterisk.org/r/3446/diff/3/?file=57809#file57809line461 Moving this to where you test bridge_channel for NULL only leaves two places where bridge_channel needs to be cleaned up instead of the current five. RAII_VAR usage could then be easily eliminated. eliminated - Jonathan --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3446/#review11736 --- On April 25, 2014, 11:54 a.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3446/ --- (Updated April 25, 2014, 11:54 a.m.) Review request for Asterisk Developers, Matt Jordan and rmudgett. Bugs: ASTERISK-23397 https://issues.asterisk.org/jira/browse/ASTERISK-23397 Repository: Asterisk Description --- r334840 removed announcements from Park manager actions back in 2011 from all of the actively supported Asterisk versions. Asterisk 12 has provided an opportunity to bring this functionality back. TimeoutChannel will now receive announcements under the strict condition that it is in a one to one bridge with Channel (the channel being parked) at the time the Park action was invoked. In this case, TimeoutChannel will be treated more or less entirely as the channel responsible for parking the call instead of just as a return point for when the call times out. Parking behavior in cases where TimeoutChannel isn't directly bridged with Channel remains mostly unchanged. The channel being parked will no longer receive announcements, but it will still be treated as having more or less self-parked. Timeout Channel will still work just as a comeback override at that point (Will be used to dial when the call times out if it's specified). AnnounceChannel field has been added to the Park action. If the AnnounceChannel field is specified and maps to an active channel, a parking announcement listener stasis subscription will be applied to that channel. When Channel is parked, that listener will trip and apply the announcement bridge feature to the AnnounceChannel. Provided that AnnounceChannel is in some kind of bridge that can use features at that point (tested with two party bridges and holding bridges), the AnnounceChannel will receive the parking announcement while staying on the bridge. If AnnounceChannel and TimeoutChannel are the same channel and that channel is bridged with Channel, a safeguard is in place to make sure multiple announcements aren't queued. In that case, AnnounceChannel is just ignored. Diffs - /branches/12/res/parking/res_parking.h 412989 /branches/12/res/parking/parking_manager.c 412989 /branches/12/res/parking/parking_bridge_features.c 412989 /branches/12/CHANGES 412989 Diff: https://reviewboard.asterisk.org/r/3446/diff/ Testing --- Tested Parking with the park action using different parking lot and timeout settings under the following scenarios: ___ Channel: SIP channel in a holding bridge TimeoutChannel: SIP channel in another holding bridge AnnounceChannel: same as TimeoutChannel Results: Timeout Channel received announcements, remained in holding bridge, and was set as the comeback dial channel. Channel gets dialed upon timeout. --- Channel: SIP channel talking to TimeoutChannel TimeoutChannel: SIP channel talking to Channel AnnounceChannel: both unspecified and the same as TimeoutChannel Results: TimeoutChannel received announcements and then hung up... treated as the Parker of the call. Gets dialed after timeout. --- Channel: Local channel in a Holding Bridge TimeoutChannel: SIP channel talking to another, unrelated SIP channel AnnounceChannel: Same as TimeoutChannel Results: TimeoutChannel receives announcements, acts as comeback dial channel. --- Channel: Local channel in a Holding Bridge TimeoutChannel: SIP channel talking to another, unrelated SIP channel AnnounceChannel: Unspecified Results: SIP channel acts as comeback dial channel, but does not receive announcements Thanks, Jonathan Rose -- _ -- 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] 3417: Add AMI events for all device state and presence state changes
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3417/#review11747 --- Ship it! /trunk/res/res_manager_devicestate.c https://reviewboard.asterisk.org/r/3417/#comment21555 Could just combine these lines: topic_forward = stasis_forward_cancel.. /trunk/res/res_manager_presencestate.c https://reviewboard.asterisk.org/r/3417/#comment21554 Same here. - Corey Farrell On April 23, 2014, 7:02 p.m., Mark Michelson wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3417/ --- (Updated April 23, 2014, 7:02 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- AMI does not emit events when device state or presence state changes. The closest things that exist currently are the ExtenstionStatus and PresenceStatus events, which inform about device state and presence state events as they pertain to hints in the dialplan. These new events are raised for every device state change or presence state change in Asterisk. Diffs - /trunk/res/res_manager_presencestate.c PRE-CREATION /trunk/res/res_manager_devicestate.c PRE-CREATION /trunk/main/presencestate.c 412583 /trunk/main/devicestate.c 412583 Diff: https://reviewboard.asterisk.org/r/3417/diff/ Testing --- See /r/3418 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] 3403: Test for channel Pickup
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3403/ --- (Updated April 25, 2014, 12:34 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 4988 Bugs: ASTERISK-23520 https://issues.asterisk.org/jira/browse/ASTERISK-23520 Repository: testsuite Description --- This test verifies that the following scenarios work for the Pickup application: 1. A channel starts to dial an IAX peer. While ringing, another channel picks up the first. 2. A channel starts to dial an IAX peer, and joins a pickup group. While ringing, another channel picks up the first from that pickup group. 3. A channel sets PICKUPMARK equal to a value and starts to dial an IAX peer. While ringing, another channel picks up the first using the PICKUPMARK method of the Pickup application. Diffs - ./asterisk/trunk/tests/apps/tests.yaml 4903 ./asterisk/trunk/tests/apps/directed_pickup/tests.yaml PRE-CREATION ./asterisk/trunk/tests/apps/directed_pickup/test-config.yaml 4903 ./asterisk/trunk/tests/apps/directed_pickup/run-test 4903 ./asterisk/trunk/tests/apps/directed_pickup/pickup_chan/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/apps/directed_pickup/pickup_chan/run-test PRE-CREATION ./asterisk/trunk/tests/apps/directed_pickup/pickup_chan/configs/ast1/iax.conf PRE-CREATION ./asterisk/trunk/tests/apps/directed_pickup/pickup_chan/configs/ast1/extensions.conf PRE-CREATION ./asterisk/trunk/tests/apps/directed_pickup/pickup/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/apps/directed_pickup/pickup/configs/ast1/iax.conf PRE-CREATION ./asterisk/trunk/tests/apps/directed_pickup/pickup/configs/ast1/extensions.conf PRE-CREATION ./asterisk/trunk/tests/apps/directed_pickup/configs/ast1/iax.conf 4903 ./asterisk/trunk/tests/apps/directed_pickup/configs/ast1/extensions.conf 4903 Diff: https://reviewboard.asterisk.org/r/3403/diff/ Testing --- Thanks, Benjamin Keith Ford -- _ -- 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] 3474: TLS and SRTP status not available in CLI
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3474/ --- (Updated April 25, 2014, 5:37 p.m.) Review request for Asterisk Developers. Changes --- Updated the patch to remove the red blob, put declaration of transport_type at the top and add the curlies, all per rmudgett. Bugs: ASTERISK-23564 https://issues.asterisk.org/jira/browse/ASTERISK-23564 Repository: Asterisk Description --- AFAICT there is no way to see from the CLI if TLS and SRTP are enabled for a channel. I asked on the ML and in #asterisk but received no answer other than that nobody knew how to get that info from the CLI. This patch shows TLS or non-TLS and SRTP or RTP. Diffs (updated) - /branches/11/channels/chan_sip.c 412921 Diff: https://reviewboard.asterisk.org/r/3474/diff/ Testing --- Testing was done on Asterisk-11.8.1 with TLS RPT, TLS SRTP, non-TLS RPT configured and a Nexus GSM using Linphone with similar configs. AFAICT the status of the channel and media was correctly reported for each scenario. Thanks, Patrick Laimbock -- _ -- 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] 3337: Code for DTLS retransmission
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3337/ --- (Updated April 25, 2014, 12:41 p.m.) Review request for Asterisk Developers. Changes --- Fixed link back to issue Bugs: ASTERISK-23649 https://issues.asterisk.org/jira/browse/ASTERISK-23649 Repository: Asterisk Description --- This patch adds the code to do the DTLS retransmissions in Asterisk. Diffs - http://svn.asterisk.org/svn/asterisk/branches/11/res/res_rtp_asterisk.c 412875 Diff: https://reviewboard.asterisk.org/r/3337/diff/ Testing --- I tested this with a basic SIPP script, which fakes a DTLS INVITE. Asterisk thinks that it is a DTLS call and inititates the DTLS handshake. SIPP doesn't respond to DTLS handshake, which causes the DTLS timeout and DTLS retransmission takes place. Thanks, Nitesh Bansal -- _ -- 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] 1803: P-Asserted-Identity Privacy - fixed behaviour - trust peer
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/1803/ --- (Updated April 25, 2014, 12:48 p.m.) Status -- This change has been discarded. Review request for Asterisk Developers. Bugs: ASTERISK-19465 https://issues.asterisk.org/jira/browse/ASTERISK-19465 Repository: Asterisk Description --- It seams that in Asterisk privacy with PAI is not implemented correctly. According to RFC 3325 when using privacy, PAI header should be set to caller num and name. The privacy is implemented by adding privacy: id header. Now when we use pai and callpres=prohib in P-Asserted-Identity header we have something which is not correct to any rfc. P-Asserted-Identity: Anonymous sip:anonymous@anonymous.invalid What my patch does: 1) adds new configurable parameter for peer - trustpeer (whether we should send privacy information to peer or not) 2) it adds Privacy header to trusted peer when PAI and CLIR is used (values id) 3) When PAI or RPID with CLIR is used and fromuser is set it is often used for authentication/recognition of the peer on the other side so we set the proper domain (not anonymous.invalid) Diffs - /trunk/configs/sip.conf.sample 358608 /trunk/channels/sip/include/sip.h 358575 /trunk/channels/chan_sip.c 358575 Diff: https://reviewboard.asterisk.org/r/1803/diff/ Testing --- I've done some basing test with outgoing calls and everything seems to works fine. Thanks, jamicque -- _ -- 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] 3474: TLS and SRTP status not available in CLI
On April 25, 2014, 1:03 p.m., rmudgett wrote: /branches/11/channels/chan_sip.c, lines 21287-21295 https://reviewboard.asterisk.org/r/3474/diff/3/?file=57909#file57909line21287 These are supposed to be AST_TRANSPORT_xxx declarations. SIP_TRANSPORT_xxx declarations don't exist. Please at least compile the patch. Heh. These were changed from SIP_TRANSPORT_xxx to AST_TRANSPORT_xxx in v12. - rmudgett --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3474/#review11751 --- On April 25, 2014, 12:37 p.m., Patrick Laimbock wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3474/ --- (Updated April 25, 2014, 12:37 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-23564 https://issues.asterisk.org/jira/browse/ASTERISK-23564 Repository: Asterisk Description --- AFAICT there is no way to see from the CLI if TLS and SRTP are enabled for a channel. I asked on the ML and in #asterisk but received no answer other than that nobody knew how to get that info from the CLI. This patch shows TLS or non-TLS and SRTP or RTP. Diffs - /branches/11/channels/chan_sip.c 412921 Diff: https://reviewboard.asterisk.org/r/3474/diff/ Testing --- Testing was done on Asterisk-11.8.1 with TLS RPT, TLS SRTP, non-TLS RPT configured and a Nexus GSM using Linphone with similar configs. AFAICT the status of the channel and media was correctly reported for each scenario. Thanks, Patrick Laimbock -- _ -- 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] 3471: Filesystem based dynamic MoH classes
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3471/#review11750 --- While I appreciate the contribution to Asterisk and the intended purpose of this patch, at this time, I don't think this patch is appropriate for inclusion. (1) Having custom file formats via the proposed playlists.txt is not something to encourage. Asterisk has an understood approach to defining its configuration; adding a custom schema creates a burden on system administrators as yet another thing to understand. Even the new configuration framework/sorcery API in Asterisk 12+ still builds on the schemas defined for .conf files. (2) By creating your own file format, you discard a substantial amount of work that has gone into the existing file reading APIs. Those APIs allow for you to check a .conf file to determine if it has changed and needs to be re-read - by rolling your own format, you are having to read the format each time Asterisk needs to determine if anything has changed in the MOH definition. While you could implement a similar mechanism, re-inventing the wheel should be a sign that something is not correct with this approach. (3) Even assuming there was a playlists.conf, I don't understand how there is a substantial benefit of having a playlists file over files in a directory. The files in the directory can already be re-scanned for new files. The files don't even have to exist in that directory: symbolic links allow for a user to have the actual files located in some other location on the server. This feels like a lot of extra work for very little benefit. Now, looking at the actual use case quoted on the issue: One of the things I needed was the ability to set the music on hold class for a call based on information gathered at the time of the call. That's fine: but how does the CHANNEL function's musicclass attribute (https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CHANNEL) not already allow for this? This patch allows us to build a MOH class and playlists on the fly that are then active when the caller puts calls on hold or if the AGI/AMI needs to play back MOH. Without this patch all combinations of music files and all MOH classes would have to be defined in the configuration file or database before Asterisk is started. In theory you could modify configuration on the fly however if I recall a reload of MOH kills other calls on hold or other nasty things happened. (1) The approach taken here gains a small amount of dynamic ability - that can mostly be captured by existing mechanisms - at a performance and maintenance cost. That's not acceptable. (2) The goal of having musiconhold 'discover' music classes at run-time is laudable, but is also possible with frameworks in Asterisk 12/trunk today. This doesn't require playlists files, a defined directory structure, or other non-standard approaches. If musiconhold was made to use sorcery, two things would be possible: (a) It would - when querying a realtime backend - grab the requested class if it existed or fail if the class did not. This would not require a reload of the module when adding a new class. (b) If using a static backend (such as a conf file), a reload would still be necessary. However, since sorcery guarantees thread safety and that existing operations in flight continue on without being affected, this would not result in any of the situations you may have run into in the past. The point is, there are mechanisms to achieve the functionality you're trying to achieve, and the approach taken here chooses not to take them. If you'd like to re-work the patch to use the proposed frameworks, we'd love to help point you in the right direction and assist with the effort. You can continue the discussion of those approaches on the asterisk-dev mailing list or in #asterisk-dev. - Matt Jordan On April 23, 2014, 9:02 a.m., Vitezslav Novy wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3471/ --- (Updated April 23, 2014, 9:02 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-23636 https://issues.asterisk.org/jira/browse/ASTERISK-23636 Repository: Asterisk Description --- This patch introduces another approach to dynamically controlled MoH. Unlike realtime this way is file based. As a switch between normal and alternative behavior, boolean variable 'dynamic' is used in MoH config file. By setting dynamic=yes new behavior is switched on. How dynamic behavior works All static MoH classes in musiconhold.conf and realtime are ignored, except [default] class. On the other hand dynamic class named
Re: [asterisk-dev] [Code Review] 3474: TLS and SRTP status not available in CLI
On April 25, 2014, 1:03 p.m., rmudgett wrote: /branches/11/channels/chan_sip.c, lines 21287-21295 https://reviewboard.asterisk.org/r/3474/diff/3/?file=57909#file57909line21287 These are supposed to be AST_TRANSPORT_xxx declarations. SIP_TRANSPORT_xxx declarations don't exist. Please at least compile the patch. rmudgett wrote: Heh. These were changed from SIP_TRANSPORT_xxx to AST_TRANSPORT_xxx in v12. We can take care of that in the merge-ness. If this is the only problem left, I'd say it's ready to go. - Matt --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3474/#review11751 --- On April 25, 2014, 12:37 p.m., Patrick Laimbock wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3474/ --- (Updated April 25, 2014, 12:37 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-23564 https://issues.asterisk.org/jira/browse/ASTERISK-23564 Repository: Asterisk Description --- AFAICT there is no way to see from the CLI if TLS and SRTP are enabled for a channel. I asked on the ML and in #asterisk but received no answer other than that nobody knew how to get that info from the CLI. This patch shows TLS or non-TLS and SRTP or RTP. Diffs - /branches/11/channels/chan_sip.c 412921 Diff: https://reviewboard.asterisk.org/r/3474/diff/ Testing --- Testing was done on Asterisk-11.8.1 with TLS RPT, TLS SRTP, non-TLS RPT configured and a Nexus GSM using Linphone with similar configs. AFAICT the status of the channel and media was correctly reported for each scenario. Thanks, Patrick Laimbock -- _ -- 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] 3474: TLS and SRTP status not available in CLI
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3474/#review11755 --- Ship it! I think this is a very good addition ready to be merged in. - Olle E Johansson On April 25, 2014, 7:37 p.m., Patrick Laimbock wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3474/ --- (Updated April 25, 2014, 7:37 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-23564 https://issues.asterisk.org/jira/browse/ASTERISK-23564 Repository: Asterisk Description --- AFAICT there is no way to see from the CLI if TLS and SRTP are enabled for a channel. I asked on the ML and in #asterisk but received no answer other than that nobody knew how to get that info from the CLI. This patch shows TLS or non-TLS and SRTP or RTP. Diffs - /branches/11/channels/chan_sip.c 412921 Diff: https://reviewboard.asterisk.org/r/3474/diff/ Testing --- Testing was done on Asterisk-11.8.1 with TLS RPT, TLS SRTP, non-TLS RPT configured and a Nexus GSM using Linphone with similar configs. AFAICT the status of the channel and media was correctly reported for each scenario. Thanks, Patrick Laimbock -- _ -- 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] 3481: Websocket: Add locking around session access and modification
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3481/ --- Review request for Asterisk Developers. Bugs: ASTERISK-23605 https://issues.asterisk.org/jira/browse/ASTERISK-23605 Repository: Asterisk Description --- This resolves a race condition where data could be written to a NULL FILE pointer causing a crash as a websocket connection was in the process of shutting down by adding locking to accesses and modifications of the websocket session struct. Diffs - branches/11/res/res_http_websocket.c 413007 Diff: https://reviewboard.asterisk.org/r/3481/diff/ Testing --- 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] 3481: Websocket: Add locking around session access and modification
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3481/#review11756 --- branches/11/res/res_http_websocket.c https://reviewboard.asterisk.org/r/3481/#comment21562 So, while locking may solve the issue, there's something more insidious about this part of the code that doesn't sit well with me. (Note: this is the actual culprit that causes a crash in the websocket write) I'm not sure that the way this is currently handled is the right way to handle a AST_WEBSOCKET_OPCODE_CLOSE. The session destructor will already close the the file descriptor. Ideally, we'd just let the destruction of the session do this work for us. It feels like the right way to handle this may be to just let the caller of ast_websocket_read know that they were told that the session needs to die. That would let them de-ref the session appropriately. If a concurrent write was occurring at the same time, when the write completes, the session would be terminated. Now, whether or not it's allowed to have a write complete when you've just been told to close the websocket is another question. If not, then we have to keep all of the locking in here. - Matt Jordan On April 25, 2014, 1:46 p.m., opticron wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3481/ --- (Updated April 25, 2014, 1:46 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-23605 https://issues.asterisk.org/jira/browse/ASTERISK-23605 Repository: Asterisk Description --- This resolves a race condition where data could be written to a NULL FILE pointer causing a crash as a websocket connection was in the process of shutting down by adding locking to accesses and modifications of the websocket session struct. Diffs - branches/11/res/res_http_websocket.c 413007 Diff: https://reviewboard.asterisk.org/r/3481/diff/ Testing --- 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] 3480: chan_pjsip: Implement get_pvt_uniqueid channel technology callback.
On April 25, 2014, 12:02 p.m., Matt Jordan wrote: /branches/12/channels/chan_pjsip.c, line 927 https://reviewboard.asterisk.org/r/3480/diff/1/?file=57904#file57904line927 I'm not sure about using threadstorage for this. One of the places that this gets called from is the bridging core via set_bridge_peer_vars_2party. That particular call can happen on a number of different threads, and will always involve callbacks into multiple channels on the same thread of execution. Joshua Colp wrote: The code actually strdupas the value in that case so it won't be a problem there. My only other options are: 1. Change that callback to return an allocated value 2. Duplicate storage of the call-id in PJSIP land and place it on the session As long as there's not a cross-thread worry, then I don't mind the thread local storage. - Matt --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3480/#review11748 --- On April 25, 2014, 11:43 a.m., Joshua Colp wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3480/ --- (Updated April 25, 2014, 11:43 a.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- This change implements the get_pvt_uniqueid channel technology callback in chan_pjsip which returns the call-id of the underlying dialog in use. Diffs - /branches/12/channels/chan_pjsip.c 413007 Diff: https://reviewboard.asterisk.org/r/3480/diff/ Testing --- Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Function Read - Timeout
Just a quick suggestion to enhance function Read. I am using function read in places to provide options to skip announcements or provide hidden features. However the minimum timeout is 1 second which puts an unnatural pause in the flow of announcements when not skipping. It would be great if there was a parameter not to wait for digits. Possible? Best regards J --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com -- _ -- 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] Function Read - Timeout
You're holding it wrong.
There are several ways to accomplish this, the easiest is to play all sound
files with one Read, like:
Read(fwdto,call-fwd-unconditionalplease-enter-thedigits/11digit/igc/sounds/destinationtelephone-number,11,,1,6)
If you can't play all the sound files with one Read, then use WaitExten and
Background:
exten = s,1,Noop(Switch Manager IVR)
same = n,Answer
same = n,Ringing
same = n,Wait(1)
same = n,Set(LOCAL(count)=0)
same = n,While($[${count} 4])
same = n,Set(count=$[${count}+1])
same = n,Background(please-enter-the/igc/sounds/destinationnumber)
same = n,WaitExten(5)
same = n,EndWhile()
same = n,Playback(goodbye)
-Original Message-
From: asterisk-dev-boun...@lists.digium.com
[mailto:asterisk-dev-boun...@lists.digium.com] On Behalf Of Jonathan White
Sent: Friday, April 25, 2014 3:12 PM
To: asterisk-dev@lists.digium.com
Subject: [asterisk-dev] Function Read - Timeout
Just a quick suggestion to enhance function Read.
I am using function read in places to provide options to skip announcements or
provide hidden features. However the minimum timeout is 1 second which puts an
unnatural pause in the flow of announcements when not skipping.
It would be great if there was a parameter not to wait for digits. Possible?
Best regards
J
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because avast! Antivirus http://www.avast.com/ protection is active.
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Re: [asterisk-dev] [Code Review] 3428: Testsuite: ARI Playback Tones tests for channels and bridges
On April 10, 2014, 6:44 a.m., Matt Jordan wrote: /asterisk/trunk/tests/rest_api/bridges/playback/tones/test-config.yaml, line 80 https://reviewboard.asterisk.org/r/3428/diff/1/?file=57149#file57149line80 This is a bridge test, so you need dependencies other than just res_ari_channels added res_ari_bridges - Jonathan --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3428/#review11540 --- On April 17, 2014, 5:21 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3428/ --- (Updated April 17, 2014, 5:21 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-23433 https://issues.asterisk.org/jira/browse/ASTERISK-23433 Repository: testsuite Description --- The YAML files have pretty apt descriptions. Channel version: * Originate a channel * Playback a tone * Pause it * Unpause it * Restart it * Delete the tone playback * Delete the channel * Validate all the events Bridge version: * Originate a channel * Create a bridge * Add the channel to the bridge * Start a tone playback on the bridge * Delete the tone playback * Delete the channel * Validate all the events Diffs - /asterisk/trunk/tests/rest_api/playback/tones/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/playback/tones/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/playback/tests.yaml 4944 /asterisk/trunk/tests/rest_api/bridges/tests.yaml 4944 /asterisk/trunk/tests/rest_api/bridges/playback/tones/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/bridges/playback/tones/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/bridges/playback/tones/bridges_play.py PRE-CREATION /asterisk/trunk/tests/rest_api/bridges/playback/tests.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/bridges/bridge_play/test-config.yaml 4944 /asterisk/trunk/tests/rest_api/bridges/bridge_play/configs/ast1/extensions.conf 4944 /asterisk/trunk/tests/rest_api/bridges/bridge_play/bridges_play.py 4944 Diff: https://reviewboard.asterisk.org/r/3428/diff/ Testing --- Ran tests, varied results, the usual. They aren't especially changed from the tests they are based on (in each case there is an existing baseline test in the same folder which handles sounds). Thanks, Jonathan Rose -- _ -- 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] 3420: Testsuite: Call Files' Max Retries
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3420/ --- (Updated April 25, 2014, 8:28 p.m.) Review request for Asterisk Developers. Changes --- Fixed the test to where I didn't need to inherit from the base CallFiles class. Bugs: ASTERISK-23218 https://issues.asterisk.org/jira/browse/ASTERISK-23218 Repository: testsuite Description --- These tests involved checking that call files max retries are functioning as planned through four tests: 1) The first test (call_file_retries_fail) required that the call file originates a local channel to a dialplan extension that will always fail, and checks to make sure that it ran through all of its max retries. 2) The second test (call_file_retries_success) involves a call file that originates a local channel that will fail once, but then is answered before it hits its max retries. 3) The third test (call_file_retries_alwaysdelete) consists of checking whether or not the call file was deleted from the [astspooldir]'s outgoing folder when the alwaysdelete option is set to 'no'. 4) The fourth and final test (call_file_retries_archive) consists of checking whether or not the call file was placed in [astspooldir]'s outgoing_done folder when archive is set to 'yes'. Diffs (updated) - ./asterisk/trunk/tests/pbx/tests.yaml 4990 ./asterisk/trunk/tests/pbx/call_file_retries_success/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_success/retries_success.py PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_success/configs/ast1/extensions.conf PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_fail/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_fail/retries_fail.py PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_fail/configs/ast1/extensions.conf PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_archive/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_archive/retries_archive.py PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_archive/configs/ast1/extensions.conf PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_alwaysdelete/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_alwaysdelete/retries_alwaysdelete.py PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_alwaysdelete/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3420/diff/ Testing --- Thanks, Scott Emidy -- _ -- 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] 3476: Memory Pre- and Post-Test Condition
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3476/#review11758 --- ./asterisk/trunk/lib/python/asterisk/memory_test_condition.py https://reviewboard.asterisk.org/r/3476/#comment21564 You may wanna change this to your tests specific functionality name/description. - Scott Emidy On April 25, 2014, 3:41 p.m., Benjamin Keith Ford wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3476/ --- (Updated April 25, 2014, 3:41 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-18429 https://issues.asterisk.org/jira/browse/ASTERISK-18429 Repository: testsuite Description --- This testcondition can be enabled for any test using the keyword 'memory' under testconditions. The purpose of this testcondition is to check the memory allocated before and after the test, and make sure they are within a certain range. If the test wants to look at something specific (such as channel.c), then each allocation that you want to look at can also be specified in under 'allocations'. If both the global memory and individual allocations are to be checked by the test, that option can be enabled by setting 'both' to value True. Diffs - ./asterisk/trunk/test-config.yaml 4944 ./asterisk/trunk/sample-yaml/memorytestcondition-config.yaml.sample PRE-CREATION ./asterisk/trunk/lib/python/asterisk/test_conditions.py 4944 ./asterisk/trunk/lib/python/asterisk/test_case.py 4944 ./asterisk/trunk/lib/python/asterisk/memory_test_condition.py PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3476/diff/ Testing --- Thanks, Benjamin Keith Ford -- _ -- 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] 3476: Memory Pre- and Post-Test Condition
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3476/ --- (Updated April 25, 2014, 8:42 p.m.) Review request for Asterisk Developers. Changes --- - Fixed a comment error Bugs: ASTERISK-18429 https://issues.asterisk.org/jira/browse/ASTERISK-18429 Repository: testsuite Description --- This testcondition can be enabled for any test using the keyword 'memory' under testconditions. The purpose of this testcondition is to check the memory allocated before and after the test, and make sure they are within a certain range. If the test wants to look at something specific (such as channel.c), then each allocation that you want to look at can also be specified in under 'allocations'. If both the global memory and individual allocations are to be checked by the test, that option can be enabled by setting 'both' to value True. Diffs (updated) - ./asterisk/trunk/test-config.yaml 4944 ./asterisk/trunk/sample-yaml/memorytestcondition-config.yaml.sample PRE-CREATION ./asterisk/trunk/lib/python/asterisk/test_conditions.py 4944 ./asterisk/trunk/lib/python/asterisk/test_case.py 4944 ./asterisk/trunk/lib/python/asterisk/memory_test_condition.py PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3476/diff/ Testing --- Thanks, Benjamin Keith Ford -- _ -- 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] 3428: Testsuite: ARI Playback Tones tests for channels and bridges
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3428/ --- (Updated April 25, 2014, 4:21 p.m.) Review request for Asterisk Developers. Changes --- * Address the findings posted by mjordan * Eliminate the need for callbacks in the bridge play tones test where I could by doing more the way the channels test was doing things * Add channels test for tones with tone zone specified in the URI Bugs: ASTERISK-23433 https://issues.asterisk.org/jira/browse/ASTERISK-23433 Repository: testsuite Description --- The YAML files have pretty apt descriptions. Channel version: * Originate a channel * Playback a tone * Pause it * Unpause it * Restart it * Delete the tone playback * Delete the channel * Validate all the events Bridge version: * Originate a channel * Create a bridge * Add the channel to the bridge * Start a tone playback on the bridge * Delete the tone playback * Delete the channel * Validate all the events Diffs (updated) - /asterisk/trunk/tests/rest_api/channels/playback/tests.yaml 4991 /asterisk/trunk/tests/rest_api/bridges/playback/tones/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/bridges/playback/tones/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/bridges/playback/tones/bridges_play.py PRE-CREATION /asterisk/trunk/tests/rest_api/bridges/playback/tests.yaml 4991 Diff: https://reviewboard.asterisk.org/r/3428/diff/ Testing --- Ran tests, varied results, the usual. They aren't especially changed from the tests they are based on (in each case there is an existing baseline test in the same folder which handles sounds). Thanks, Jonathan Rose -- _ -- 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] 3482: func_presencestate: Make base64 encoded-ness consistent for consumers of presence state
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3482/
---
Review request for Asterisk Developers.
Bugs: ASTERISK-23671
https://issues.asterisk.org/jira/browse/ASTERISK-23671
Repository: Asterisk
Description
---
The 'e' option for the PRESENCE_STATE() function is not very well defined.
Specifically, when using the function in write mode, it is unclear whether
consumers of presence state events should expect to receive base64-encoded
values or not. Further, the behavior is not consistent within the module. When
the initial presence state is written, base64-encoded values are written to
stasis and consumers receive these encoded values. However, if the
ast_presence_state() function is called to retrieve the current presence
values, decoded values are returned.
With this patch, if the subtype and message given in the PRESENCE_STATE()
function are base64-encoded, these values are decoded before being sent to
stasis. This way, consumers of presence state will always be guaranteed to get
decoded values.
So with this patch, you can do the following:
exten =
blah,1,Set(PRESENCE_STATE(CustomPresence:blah)=away,bHVuY2g=,Q2xlbSdzIENsYW1z,e)
; Sends consumers state=away, subtype=lunch, message=Clem's Clams. Stores
base64 in astdb
exten = blah,n,Set(subtype=${PRESENCE_STATE(CustomPresence:blah,subtype)}) ;
Sets subtype to lunch
exten = blah,n,Set(message=${PRESENCE_STATE(CustomPresence:blah,message)}) ;
Sets message to Clem's Clams
If you actually want to be sending Base64-encoded data to consumers, then omit
the e option.
exten =
blah,1,Set(PRESENCE_STATE(CustomPresence:blah)=away,bHVuY2g=,Q2xlbSdzIENsYW1z)
; Sends consumers state=away, subtype=bHVuY2g=, message=Q2xlbSdzIENsYW1z.
Stores base64 in astdb
exten = blah,n,Set(subtype=${PRESENCE_STATE(CustomPresence:blah,subtype)}) ;
Sets subtype to bHVuY2g=
exten = blah,n,Set(message=${PRESENCE_STATE(CustomPresence:blah,message)}) ;
Sets message to Q2xlbSdzIENsYW1z
exten =
blah,n,Set(subtype=${BASE64_DECODE(${PRESENCE_STATE(CustomPresence:blah,subtype)})})
; Sets subtype to lunch
exten =
blah,n,Set(message=${BASE64_DECODe(${PRESENCE_STATE(CustomPresence:blah,message)})})
; Sets message to Clem's Clams
To me, this behavior seems at the very least more consistent than what was
being done before. I'm certainly willing to hear objections, though.
Diffs
-
/trunk/funcs/func_presencestate.c 412583
Diff: https://reviewboard.asterisk.org/r/3482/diff/
Testing
---
I have added a unit test that ensures this behaves as expected. In doing so, I
realized it was nearly identical to the previous test_presence_state_change
test, so I refactored the code to be reusable and to plug some memory leaks and
stasis subscription leaks.
Thanks,
Mark Michelson
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[asterisk-dev] [Code Review] 3485: pjsip: Fix a bug where transferring to a parking extension causes calls to hang
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3485/ --- Review request for Asterisk Developers, Matt Jordan and Mark Michelson. Repository: Asterisk Description --- If a PJSIP endpoint attempts to blind transfer to a parking extension, there is an override to the normal transfer logic that can make things act a little weird. We noticed that this would leave various phones hanging on transfer screens without progressing. When the transfer was considered successful, PJSIP deferred the actual action of sending the 200 notify and the actual trigger for that happening never occurs when the transfer is to a parking extension. In order to handle this, the bridge function that handles blind transfers now returns a different value if a call was parked and if the channel driver needs to react differently in this case, it can. In the case of PJSIP, we respond to transfers to park by immediately sending the notify with 200 OK sip frag instead of deferring the action. Diffs - /branches/12/res/res_pjsip_refer.c 412824 /branches/12/main/manager.c 412824 /branches/12/main/bridge_basic.c 412824 /branches/12/main/bridge.c 412824 /branches/12/include/asterisk/bridge.h 412824 /branches/12/channels/chan_unistim.c 412824 /branches/12/channels/chan_skinny.c 412824 /branches/12/channels/chan_sip.c 412824 /branches/12/channels/chan_oss.c 412824 /branches/12/channels/chan_iax2.c 412824 Diff: https://reviewboard.asterisk.org/r/3485/diff/ Testing --- Before patch: * Blind transfer on Polycom SPIP: Phone is on the blind transfer screen until it either manually hangs up or 60 seconds pass and Asterisk terminates the session. After the patch: * Blind transfer on Polycom SPIP: Phone immediately leaves the blind transfer screen and goes back to idle mode. Thanks, Jonathan Rose -- _ -- 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] Function Read - Timeout
Yes that's a good idea. This fixes one of my issues however it doesn't when
I have two reads one after the other.
It would still be good if there was a parameter to have no delay.
J
-Original Message-
From: Eric Wieling
Sent: Friday, April 25, 2014 8:54 PM
To: Jonathan White ; Asterisk Developers Mailing List
Subject: RE: [asterisk-dev] Function Read - Timeout
You're holding it wrong.
There are several ways to accomplish this, the easiest is to play all sound
files with one Read, like:
Read(fwdto,call-fwd-unconditionalplease-enter-thedigits/11digit/igc/sounds/destinationtelephone-number,11,,1,6)
If you can't play all the sound files with one Read, then use WaitExten and
Background:
exten = s,1,Noop(Switch Manager IVR)
same = n,Answer
same = n,Ringing
same = n,Wait(1)
same = n,Set(LOCAL(count)=0)
same = n,While($[${count} 4])
same = n,Set(count=$[${count}+1])
same = n,Background(please-enter-the/igc/sounds/destinationnumber)
same = n,WaitExten(5)
same = n,EndWhile()
same = n,Playback(goodbye)
-Original Message-
From: asterisk-dev-boun...@lists.digium.com
[mailto:asterisk-dev-boun...@lists.digium.com] On Behalf Of Jonathan White
Sent: Friday, April 25, 2014 3:12 PM
To: asterisk-dev@lists.digium.com
Subject: [asterisk-dev] Function Read - Timeout
Just a quick suggestion to enhance function Read.
I am using function read in places to provide options to skip announcements
or provide hidden features. However the minimum timeout is 1 second which
puts an unnatural pause in the flow of announcements when not skipping.
It would be great if there was a parameter not to wait for digits. Possible?
Best regards
J
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Re: [asterisk-dev] asterisk-dev Digest, Vol 117, Issue 173
Hi White, It is no problem. This is a small function which is customized by myself for internal calls only in my company. It is not commercial activity. So , it is not something violation. Best regards, NHSON -Original Message- From: asterisk-dev-boun...@lists.digium.com [mailto:asterisk-dev-boun...@lists.digium.com] On Behalf Of asterisk-dev-requ...@lists.digium.com Sent: Friday, April 25, 2014 9:08 PM To: asterisk-dev@lists.digium.com Subject: asterisk-dev Digest, Vol 117, Issue 173 Send asterisk-dev mailing list submissions to asterisk-dev@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-dev or, via email, send a message with subject or body 'help' to asterisk-dev-requ...@lists.digium.com You can reach the person managing the list at asterisk-dev-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-dev digest... Today's Topics: 1. Re: Add new option to Queue function (jonathan white) 2. Re: [Code Review] 3479: chan_pjsip: Call pickup test. (Matt Jordan) -- Message: 1 Date: Fri, 25 Apr 2014 15:15:20 +0100 From: jonathan white j...@uvacity.com To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com Subject: Re: [asterisk-dev] Add new option to Queue function Message-ID: cac5vygnx_ckz+yyud_ghcthr2fywbe9kl1tbfejbx9mzzxp...@mail.gmail.com Content-Type: text/plain; charset=utf-8 Just something I know which may restrict what can be done. Avaya have many patents for call distribution. This includes call distribution to agents who have spent the least amount of time on the phone and taken the lowest number of calls. On 25 Apr 2014 15:00, Nguyen Hoang Son nh...@vasc.com.vn wrote: Hi all, I'm using Queue function of Asterisk to arrange calls which is coming to my agents. I want to customize the way asterisk arrange coming call, in other word, is it possible to create a new option instead of using the existing: RINGALL, ROUNDROBIN,... . For example: The incoming call should come to the argent who has the most waiting time (idle time). I find out that the algorithm of each option of Queue is defined in app_queue.c in the source code but I don't know how to change, how to add the waiting time as a new option to sort by. This question is quite related to the development of asterisk, so please help if you have any idea or experience on that. Thank you very much. --- *NGUY?N HO?NG S?N* M-Commerce Center VASC Software and Media Company - VNPT Addr: No.97 Nguyen Chi Thanh Street, Dong Da District, Hanoi, Vietnam Cell phone: +84 912998101 Skype: hoangsonk49 E-mail: nh...@vasc.com.vn -- _ -- 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 -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20140425/c210fe8 d/attachment-0001.html -- Message: 2 Date: Fri, 25 Apr 2014 14:19:55 - From: Matt Jordan reviewbo...@asterisk.org To: Joshua Colp jc...@digium.com, Asterisk Developers asterisk-dev@lists.digium.com, Matt Jordan reviewbo...@asterisk.org Subject: Re: [asterisk-dev] [Code Review] 3479: chan_pjsip: Call pickup test. Message-ID: 20140425141955.11866.65...@sonic.digium.api Content-Type: text/plain; charset=utf-8 --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3479/#review11742 --- /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test https://reviewboard.asterisk.org/r/3479/#comment21536 2014 /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test https://reviewboard.asterisk.org/r/3479/#comment21535 Are you sure you're Jonathan Rose? /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test https://reviewboard.asterisk.org/r/3479/#comment21543 These are always used as regular expressions. Why not just compile them here and use them as such everywhere else? /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test https://reviewboard.asterisk.org/r/3479/#comment21537 Since this is using PJSIP, there's no need to support previous versions of Asterisk. Just the bridging model for 12 is sufficient. /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test https://reviewboard.asterisk.org/r/3479/#comment21540 And just use 12 here as well /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test https
