Re: [asterisk-dev] [Code Review] 4597: res_pjsip: add CLI command to show global and system configuration
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4597/#review15148 --- Ship it! Ran the command, format looks good - all the settings show up and reflect accurately. Changed settings, I see the new settings after restart. - rnewton On April 8, 2015, 5:52 p.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4597/ --- (Updated April 8, 2015, 5:52 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24918 https://issues.asterisk.org/jira/browse/ASTERISK-24918 Repository: Asterisk Description --- Added a new CLI command for res_pjsip that shows both global and system configuration settings: pjsip show settings Diffs - branches/13/res/res_pjsip/include/res_pjsip_private.h 434423 branches/13/res/res_pjsip/config_system.c 434423 branches/13/res/res_pjsip/config_global.c 434423 branches/13/res/res_pjsip.c 434423 branches/13/CHANGES 434423 Diff: https://reviewboard.asterisk.org/r/4597/diff/ Testing --- Ran the command and checked output. Changed some options and reloaded and made sure global settings changed, but system ones did not. Changed some settings again and restarted and made sure both global and system changes took effect. Also removed the sections completely from the pjsip.conf file and made sure the defaults were shown. 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] 4588: IAX make calltoken expiration time configurable
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4588/#review15050 --- You probably want to add documentation for the new configuration option to any appropriate place, such as the iax.conf.sample file. - rnewton On April 3, 2015, 9:32 p.m., Y Ateya wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4588/ --- (Updated April 3, 2015, 9:32 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24939 https://issues.asterisk.org/jira/browse/ASTERISK-24939 Repository: Asterisk Description --- The section 4.1 in call token changes to IAX protocol (http://downloads.asterisk.org/pub/security/IAX2-security.html): The token timeout will be hard coded at 10 seconds for now. However, it may be made configurable at some point if it seems to be a useful addition In case of lagged network cases (or bad network which required multiple retries) 10 seconds is not enough. Diffs - trunk/channels/chan_iax2.c 432806 Diff: https://reviewboard.asterisk.org/r/4588/diff/ Testing --- Thanks, Y Ateya -- _ -- 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] 4488: Super Awesome Company: Phase 1 - Patch 2 - Outside Connectivity!
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4488/ --- (Updated March 24, 2015, 9:53 p.m.) Review request for Asterisk Developers. Changes --- Addressed all findings, modified a few comments here or there. Tested with 4503 and 4504. A quick summary of the testing I ran through. I tested all the requirements for all four patches combined. *patch1 - internal stuff* internal user to internal user (audio): PASS internal user to internal user (voicemail-unavail): PASS internal user to internal user (voicemail-busy): PASS internal user can check voicemail: PASS deskphone displays MWI indication: PASS *patch2 - outside connectivity* registration to ITSP(DCS) comes up: PASS internal user dials out ITSP with 7 digit number: PASS internal user dials out ITSP with 10 digit number: PASS internal user dials out ITSP with 10+1 digit number: PASS internal user dials main IVR internally: PASS restricted number patterns work successfully: PASS inbound calls reach the main IVR: PASS external user can reach external voicemail feature via DID: PASS external users can dial internal users directly via DID match: PASS *patch3 - queues with external and internal access* sales queue reached internally: PASS externally: PASS sales queue rings Terry, Garnet and Franny in ring-all: PASS customer advocate queue reached internally: PASS externally: PASS customer advocate Queue rings Maria, Dusty and Tommie in ring-all: PASS *patch4 - conferences* employee conference can be dialed by internal users: PASS at least two parties in employee conference with audio: PASS customer conference can be dialed into by internal user and transfer in external users: PASS at least two parties, including an external party in customer conference with audio: PASS *ALL PATCHES COMBINED* All IVR options go to the correct feature/extension: PASS CDR Master.csv does not record any intra-office calls: PASS CDR Master.csv records calls to/from the ITSP account: PASS Repository: Asterisk Description --- Howdy, here is another patch for the Super Awesome Company configuration. We are still in phase 1. The general requirements are posted on the wiki: https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company The specific requirements this patch meets are below: pjsip.conf * SIP ITSP configuration example and have place holders for the required authentication bits. ** Assume that Asterisk does not have a public IP address, and sits behind a NAT with its desk phones. * Have outbound registration to the SIP trunk, and an endpoint that represents the SIP trunk. * Inbound calls received from the SIP trunk should go into their own context. extensions.conf * Match the outbound dial request so that it can only dial US area codes. ** Don't let people dial 900 numbers, international numbers, or any other numbers that could result in a charge * Inbound calls from the SIP trunk should hit a basic Auto Attendant that prompts them for the extension to dial, after greeting them to SAC. * If an inbound call matches a DID that maps to a specific extension/device, dial that extension/device directly. Billing * Make sure CDRs output all calls that are from/to the SIP trunk. These should be logged to a CSV. * For intra-office calls, kill the CDRs. Additional Requirements Noted: * For outbound calls, each SAC employee’s 10-digit DID number is provided as their Caller ID. * Voicemail may be accessed remotely by employees who dial 256-555-1234. When employees dial voicemail remotely, they must input both their mailbox number and their pin code. * 7, 10 and 10+1 digit dialing for local and long distance calls. * Internal dialing of otherwise inbound features, ** 1100 to reach the main IVR. * The IVR options possible without getting into Phase 2. Diffs (updated) - /branches/13/configs/basic-pbx/pjsip.conf 43 /branches/13/configs/basic-pbx/modules.conf 43 /branches/13/configs/basic-pbx/logger.conf 43 /branches/13/configs/basic-pbx/extensions.conf 43 /branches/13/configs/basic-pbx/cdr_custom.conf PRE-CREATION /branches/13/configs/basic-pbx/cdr.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4488/diff/ Testing --- Setup with a Digium Cloud Services trunk and a few internal phones. Internal to Internal calls. Calls Internal to voicemail and other features. External to internal DID calls. External to internal feature calls. Basically tried to call as many ways as I could through all the various features. Everything seemed to work. Thanks, rnewton -- _ -- 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
Re: [asterisk-dev] [Code Review] 4504: SAC: Add conferences for employees / employees+customers
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4504/#review14822 --- /branches/13/configs/basic-pbx/extensions.conf https://reviewboard.asterisk.org/r/4504/#comment25421 SAC doesn't have a DID for this. The reference also doesn't mention requiring external access to it. I believe the idea is that the conference is there for an employee to transfer a customer into if needed. /branches/13/configs/basic-pbx/modules.conf https://reviewboard.asterisk.org/r/4504/#comment25422 Make sure to add a timing interface, we could use res_timing_timerfd.so - rnewton On March 16, 2015, 5:48 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4504/ --- (Updated March 16, 2015, 5:48 p.m.) Review request for Asterisk Developers and rnewton. Repository: Asterisk Description --- From: https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company SAC requires two conference rooms, one for use by employees only and one for use by employees and customers (outside connectivity still needs to be established so that 555-6500 can be added and customers can actually dial into said conference) Diffs - /branches/13/configs/basic-pbx/modules.conf 432996 /branches/13/configs/basic-pbx/extensions.conf 432996 /branches/13/configs/basic-pbx/confbridge.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4504/diff/ Testing --- Made sure app_confbridge loaded and internal users were able to dial into the conferences. 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] 4504: SAC: Add conferences for employees / employees+customers
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4504/#review14821 --- Tested with 4488. Pretty much worked fine. You will need to add a timing interface to modules.conf. res_timing_timerfd.so is probably fine. - rnewton On March 16, 2015, 5:48 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4504/ --- (Updated March 16, 2015, 5:48 p.m.) Review request for Asterisk Developers and rnewton. Repository: Asterisk Description --- From: https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company SAC requires two conference rooms, one for use by employees only and one for use by employees and customers (outside connectivity still needs to be established so that 555-6500 can be added and customers can actually dial into said conference) Diffs - /branches/13/configs/basic-pbx/modules.conf 432996 /branches/13/configs/basic-pbx/extensions.conf 432996 /branches/13/configs/basic-pbx/confbridge.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4504/diff/ Testing --- Made sure app_confbridge loaded and internal users were able to dial into the conferences. 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] 4503: SAC: Configure customer advocate/sales queues
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4503/#review14823 --- /trunk/configs/basic-pbx/extensions.conf https://reviewboard.asterisk.org/r/4503/#comment25423 These should move to the External-Features context and get their extensions changed to the full DID. The Features context will have the internal goto calls to send internal callers to the extensions in External-Features. You'll note this in the diff I just E-mailed over to you. - rnewton On March 16, 2015, 3:37 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4503/ --- (Updated March 16, 2015, 3:37 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- As described on the wiki at: The Sales Queue may be reached externally by dialing 256-555-1200, or internally by dialing 1200. The Customer Experience Queue may be reached externally by dialing 256-555-1250, or internally by dialing 1250. Sales Queue Calls to the Sales Queue should ring Terry, Garnet and Franny in ring-all fashion If no one answers a call to the Sales Queue after 5 minutes, the caller should be directed to the Operator so that the Operator can take a message and have a Sales person contact the caller at a later time. Customer Advocate Queue Calls to the Customer Advocate Queue should ring Maria, Dusty and Tommie in ring-all fashion. If no one answers a call to the Customer Advocate queue after 20 minutes, the caller should be directed to the Operator so that the Operator can take a message and have a Customer Advocate contact the caller at a later time. NOTE: This review is accidentally against trunk, but the patch is perfectly portable, so that shouldn't be a problem. NOTE 2: External extensions for the queues still need to be added since the outside connectivity patch hasn't been merged yet https://reviewboard.asterisk.org/r/4488/diff/#index_header Diffs - /trunk/configs/basic-pbx/queues.conf PRE-CREATION /trunk/configs/basic-pbx/modules.conf 432443 /trunk/configs/basic-pbx/extensions.conf 432443 Diff: https://reviewboard.asterisk.org/r/4503/diff/ Testing --- Had some phones register as agents in the queues as well as the operator, made sure the queue members were dialed in ring-all fashion and that the timeouts occurred and the operator was dialed as expected. 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] 4503: SAC: Configure customer advocate/sales queues
On March 24, 2015, 10:39 p.m., rnewton wrote: /trunk/configs/basic-pbx/extensions.conf, lines 16-30 https://reviewboard.asterisk.org/r/4503/diff/1/?file=72533#file72533line16 These should move to the External-Features context and get their extensions changed to the full DID. The Features context will have the internal goto calls to send internal callers to the extensions in External-Features. You'll note this in the diff I just E-mailed over to you. The diff should also show modified comments for all of those extensions I believe. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4503/#review14823 --- On March 16, 2015, 3:37 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4503/ --- (Updated March 16, 2015, 3:37 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- As described on the wiki at: The Sales Queue may be reached externally by dialing 256-555-1200, or internally by dialing 1200. The Customer Experience Queue may be reached externally by dialing 256-555-1250, or internally by dialing 1250. Sales Queue Calls to the Sales Queue should ring Terry, Garnet and Franny in ring-all fashion If no one answers a call to the Sales Queue after 5 minutes, the caller should be directed to the Operator so that the Operator can take a message and have a Sales person contact the caller at a later time. Customer Advocate Queue Calls to the Customer Advocate Queue should ring Maria, Dusty and Tommie in ring-all fashion. If no one answers a call to the Customer Advocate queue after 20 minutes, the caller should be directed to the Operator so that the Operator can take a message and have a Customer Advocate contact the caller at a later time. NOTE: This review is accidentally against trunk, but the patch is perfectly portable, so that shouldn't be a problem. NOTE 2: External extensions for the queues still need to be added since the outside connectivity patch hasn't been merged yet https://reviewboard.asterisk.org/r/4488/diff/#index_header Diffs - /trunk/configs/basic-pbx/queues.conf PRE-CREATION /trunk/configs/basic-pbx/modules.conf 432443 /trunk/configs/basic-pbx/extensions.conf 432443 Diff: https://reviewboard.asterisk.org/r/4503/diff/ Testing --- Had some phones register as agents in the queues as well as the operator, made sure the queue members were dialed in ring-all fashion and that the timeouts occurred and the operator was dialed as expected. 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] 4488: Super Awesome Company: Phase 1 - Patch 2 - Outside Connectivity!
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4488/#review14818 --- Forgot to add that I fixed a few issues found during testing, but you'll see them in the review. - rnewton On March 24, 2015, 9:53 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4488/ --- (Updated March 24, 2015, 9:53 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- Howdy, here is another patch for the Super Awesome Company configuration. We are still in phase 1. The general requirements are posted on the wiki: https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company The specific requirements this patch meets are below: pjsip.conf * SIP ITSP configuration example and have place holders for the required authentication bits. ** Assume that Asterisk does not have a public IP address, and sits behind a NAT with its desk phones. * Have outbound registration to the SIP trunk, and an endpoint that represents the SIP trunk. * Inbound calls received from the SIP trunk should go into their own context. extensions.conf * Match the outbound dial request so that it can only dial US area codes. ** Don't let people dial 900 numbers, international numbers, or any other numbers that could result in a charge * Inbound calls from the SIP trunk should hit a basic Auto Attendant that prompts them for the extension to dial, after greeting them to SAC. * If an inbound call matches a DID that maps to a specific extension/device, dial that extension/device directly. Billing * Make sure CDRs output all calls that are from/to the SIP trunk. These should be logged to a CSV. * For intra-office calls, kill the CDRs. Additional Requirements Noted: * For outbound calls, each SAC employee’s 10-digit DID number is provided as their Caller ID. * Voicemail may be accessed remotely by employees who dial 256-555-1234. When employees dial voicemail remotely, they must input both their mailbox number and their pin code. * 7, 10 and 10+1 digit dialing for local and long distance calls. * Internal dialing of otherwise inbound features, ** 1100 to reach the main IVR. * The IVR options possible without getting into Phase 2. Diffs - /branches/13/configs/basic-pbx/pjsip.conf 43 /branches/13/configs/basic-pbx/modules.conf 43 /branches/13/configs/basic-pbx/logger.conf 43 /branches/13/configs/basic-pbx/extensions.conf 43 /branches/13/configs/basic-pbx/cdr_custom.conf PRE-CREATION /branches/13/configs/basic-pbx/cdr.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4488/diff/ Testing --- Setup with a Digium Cloud Services trunk and a few internal phones. Internal to Internal calls. Calls Internal to voicemail and other features. External to internal DID calls. External to internal feature calls. Basically tried to call as many ways as I could through all the various features. Everything seemed to work. Thanks, rnewton -- _ -- 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] 4503: SAC: Configure customer advocate/sales queues
On March 24, 2015, 10:41 p.m., rnewton wrote: Tested with 4488, only a few modifications made. Emailed you a diff with the changes and external extensions. When testing this patch with 4488 I ran through the following tests: *patch1 - internal stuff* internal user to internal user (audio): PASS internal user to internal user (voicemail-unavail): PASS internal user to internal user (voicemail-busy): PASS internal user can check voicemail: PASS deskphone displays MWI indication: PASS *patch2 - outside connectivity* registration to ITSP(DCS) comes up: PASS internal user dials out ITSP with 7 digit number: PASS internal user dials out ITSP with 10 digit number: PASS internal user dials out ITSP with 10+1 digit number: PASS internal user dials main IVR internally: PASS restricted number patterns work successfully: PASS inbound calls reach the main IVR: PASS external user can reach external voicemail feature via DID: PASS external users can dial internal users directly via DID match: PASS *patch3 - queues with external and internal access* sales queue reached internally: PASS externally: PASS sales queue rings Terry, Garnet and Franny in ring-all: PASS customer advocate queue reached internally: PASS externally: PASS customer advocate Queue rings Maria, Dusty and Tommie in ring-all: PASS *patch4 - conferences* employee conference can be dialed by internal users: PASS at least two parties in employee conference with audio: PASS customer conference can be dialed into by internal user and transfer in external users: PASS at least two parties, including an external party in customer conference with audio: PASS *ALL PATCHES COMBINED* All IVR options go to the correct feature/extension: PASS CDR Master.csv does not record any intra-office calls: PASS CDR Master.csv records calls to/from the ITSP account: PASS - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4503/#review14825 --- On March 16, 2015, 3:37 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4503/ --- (Updated March 16, 2015, 3:37 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- As described on the wiki at: The Sales Queue may be reached externally by dialing 256-555-1200, or internally by dialing 1200. The Customer Experience Queue may be reached externally by dialing 256-555-1250, or internally by dialing 1250. Sales Queue Calls to the Sales Queue should ring Terry, Garnet and Franny in ring-all fashion If no one answers a call to the Sales Queue after 5 minutes, the caller should be directed to the Operator so that the Operator can take a message and have a Sales person contact the caller at a later time. Customer Advocate Queue Calls to the Customer Advocate Queue should ring Maria, Dusty and Tommie in ring-all fashion. If no one answers a call to the Customer Advocate queue after 20 minutes, the caller should be directed to the Operator so that the Operator can take a message and have a Customer Advocate contact the caller at a later time. NOTE: This review is accidentally against trunk, but the patch is perfectly portable, so that shouldn't be a problem. NOTE 2: External extensions for the queues still need to be added since the outside connectivity patch hasn't been merged yet https://reviewboard.asterisk.org/r/4488/diff/#index_header Diffs - /trunk/configs/basic-pbx/queues.conf PRE-CREATION /trunk/configs/basic-pbx/modules.conf 432443 /trunk/configs/basic-pbx/extensions.conf 432443 Diff: https://reviewboard.asterisk.org/r/4503/diff/ Testing --- Had some phones register as agents in the queues as well as the operator, made sure the queue members were dialed in ring-all fashion and that the timeouts occurred and the operator was dialed as expected. 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] 4504: SAC: Add conferences for employees / employees+customers
On March 24, 2015, 10:32 p.m., rnewton wrote: Tested with 4488. Pretty much worked fine. You will need to add a timing interface to modules.conf. res_timing_timerfd.so is probably fine. When testing with 4488 I ran through the following tests: *patch1 - internal stuff* internal user to internal user (audio): PASS internal user to internal user (voicemail-unavail): PASS internal user to internal user (voicemail-busy): PASS internal user can check voicemail: PASS deskphone displays MWI indication: PASS *patch2 - outside connectivity* registration to ITSP(DCS) comes up: PASS internal user dials out ITSP with 7 digit number: PASS internal user dials out ITSP with 10 digit number: PASS internal user dials out ITSP with 10+1 digit number: PASS internal user dials main IVR internally: PASS restricted number patterns work successfully: PASS inbound calls reach the main IVR: PASS external user can reach external voicemail feature via DID: PASS external users can dial internal users directly via DID match: PASS *patch3 - queues with external and internal access* sales queue reached internally: PASS externally: PASS sales queue rings Terry, Garnet and Franny in ring-all: PASS customer advocate queue reached internally: PASS externally: PASS customer advocate Queue rings Maria, Dusty and Tommie in ring-all: PASS *patch4 - conferences* employee conference can be dialed by internal users: PASS at least two parties in employee conference with audio: PASS customer conference can be dialed into by internal user and transfer in external users: PASS at least two parties, including an external party in customer conference with audio: PASS *ALL PATCHES COMBINED* All IVR options go to the correct feature/extension: PASS CDR Master.csv does not record any intra-office calls: PASS CDR Master.csv records calls to/from the ITSP account: PASS - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4504/#review14821 --- On March 16, 2015, 5:48 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4504/ --- (Updated March 16, 2015, 5:48 p.m.) Review request for Asterisk Developers and rnewton. Repository: Asterisk Description --- From: https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company SAC requires two conference rooms, one for use by employees only and one for use by employees and customers (outside connectivity still needs to be established so that 555-6500 can be added and customers can actually dial into said conference) Diffs - /branches/13/configs/basic-pbx/modules.conf 432996 /branches/13/configs/basic-pbx/extensions.conf 432996 /branches/13/configs/basic-pbx/confbridge.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4504/diff/ Testing --- Made sure app_confbridge loaded and internal users were able to dial into the conferences. 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] 4503: SAC: Configure customer advocate/sales queues
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4503/#review14825 --- Tested with 4488, only a few modifications made. Emailed you a diff with the changes and external extensions. - rnewton On March 16, 2015, 3:37 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4503/ --- (Updated March 16, 2015, 3:37 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- As described on the wiki at: The Sales Queue may be reached externally by dialing 256-555-1200, or internally by dialing 1200. The Customer Experience Queue may be reached externally by dialing 256-555-1250, or internally by dialing 1250. Sales Queue Calls to the Sales Queue should ring Terry, Garnet and Franny in ring-all fashion If no one answers a call to the Sales Queue after 5 minutes, the caller should be directed to the Operator so that the Operator can take a message and have a Sales person contact the caller at a later time. Customer Advocate Queue Calls to the Customer Advocate Queue should ring Maria, Dusty and Tommie in ring-all fashion. If no one answers a call to the Customer Advocate queue after 20 minutes, the caller should be directed to the Operator so that the Operator can take a message and have a Customer Advocate contact the caller at a later time. NOTE: This review is accidentally against trunk, but the patch is perfectly portable, so that shouldn't be a problem. NOTE 2: External extensions for the queues still need to be added since the outside connectivity patch hasn't been merged yet https://reviewboard.asterisk.org/r/4488/diff/#index_header Diffs - /trunk/configs/basic-pbx/queues.conf PRE-CREATION /trunk/configs/basic-pbx/modules.conf 432443 /trunk/configs/basic-pbx/extensions.conf 432443 Diff: https://reviewboard.asterisk.org/r/4503/diff/ Testing --- Had some phones register as agents in the queues as well as the operator, made sure the queue members were dialed in ring-all fashion and that the timeouts occurred and the operator was dialed as expected. 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] 4503: SAC: Configure customer advocate/sales queues
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4503/#review14752 --- As an update - when testing with my external connectivity patch I've run into some one-way audio and issues where in certain calling scenarios MOH does not play. I'm investigating that further tomorrow. - rnewton On March 16, 2015, 3:37 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4503/ --- (Updated March 16, 2015, 3:37 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- As described on the wiki at: The Sales Queue may be reached externally by dialing 256-555-1200, or internally by dialing 1200. The Customer Experience Queue may be reached externally by dialing 256-555-1250, or internally by dialing 1250. Sales Queue Calls to the Sales Queue should ring Terry, Garnet and Franny in ring-all fashion If no one answers a call to the Sales Queue after 5 minutes, the caller should be directed to the Operator so that the Operator can take a message and have a Sales person contact the caller at a later time. Customer Advocate Queue Calls to the Customer Advocate Queue should ring Maria, Dusty and Tommie in ring-all fashion. If no one answers a call to the Customer Advocate queue after 20 minutes, the caller should be directed to the Operator so that the Operator can take a message and have a Customer Advocate contact the caller at a later time. NOTE: This review is accidentally against trunk, but the patch is perfectly portable, so that shouldn't be a problem. NOTE 2: External extensions for the queues still need to be added since the outside connectivity patch hasn't been merged yet https://reviewboard.asterisk.org/r/4488/diff/#index_header Diffs - /trunk/configs/basic-pbx/queues.conf PRE-CREATION /trunk/configs/basic-pbx/modules.conf 432443 /trunk/configs/basic-pbx/extensions.conf 432443 Diff: https://reviewboard.asterisk.org/r/4503/diff/ Testing --- Had some phones register as agents in the queues as well as the operator, made sure the queue members were dialed in ring-all fashion and that the timeouts occurred and the operator was dialed as expected. 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] 4503: SAC: Configure customer advocate/sales queues
On March 17, 2015, 9:44 p.m., Mark Michelson wrote: Everything looks good here. Should this review stay open for the external extensions to be added, or will that be a separate review? Jonathan Rose wrote: I'll leave both reviews open for now. I'm finally testing with these with outside connectivity this morning, so we'll see if anything needs to change. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4503/#review14730 --- On March 16, 2015, 3:37 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4503/ --- (Updated March 16, 2015, 3:37 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- As described on the wiki at: The Sales Queue may be reached externally by dialing 256-555-1200, or internally by dialing 1200. The Customer Experience Queue may be reached externally by dialing 256-555-1250, or internally by dialing 1250. Sales Queue Calls to the Sales Queue should ring Terry, Garnet and Franny in ring-all fashion If no one answers a call to the Sales Queue after 5 minutes, the caller should be directed to the Operator so that the Operator can take a message and have a Sales person contact the caller at a later time. Customer Advocate Queue Calls to the Customer Advocate Queue should ring Maria, Dusty and Tommie in ring-all fashion. If no one answers a call to the Customer Advocate queue after 20 minutes, the caller should be directed to the Operator so that the Operator can take a message and have a Customer Advocate contact the caller at a later time. NOTE: This review is accidentally against trunk, but the patch is perfectly portable, so that shouldn't be a problem. NOTE 2: External extensions for the queues still need to be added since the outside connectivity patch hasn't been merged yet https://reviewboard.asterisk.org/r/4488/diff/#index_header Diffs - /trunk/configs/basic-pbx/queues.conf PRE-CREATION /trunk/configs/basic-pbx/modules.conf 432443 /trunk/configs/basic-pbx/extensions.conf 432443 Diff: https://reviewboard.asterisk.org/r/4503/diff/ Testing --- Had some phones register as agents in the queues as well as the operator, made sure the queue members were dialed in ring-all fashion and that the timeouts occurred and the operator was dialed as expected. 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] 4504: SAC: Add conferences for employees / employees+customers
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4504/#review14741 --- Looks good. I'm finally testing with these with outside connectivity this morning, so we'll see if anything needs to change. - rnewton On March 16, 2015, 5:48 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4504/ --- (Updated March 16, 2015, 5:48 p.m.) Review request for Asterisk Developers and rnewton. Repository: Asterisk Description --- From: https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company SAC requires two conference rooms, one for use by employees only and one for use by employees and customers (outside connectivity still needs to be established so that 555-6500 can be added and customers can actually dial into said conference) Diffs - /branches/13/configs/basic-pbx/modules.conf 432996 /branches/13/configs/basic-pbx/extensions.conf 432996 /branches/13/configs/basic-pbx/confbridge.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4504/diff/ Testing --- Made sure app_confbridge loaded and internal users were able to dial into the conferences. 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] 4488: Super Awesome Company: Phase 1 - Patch 2 - Outside Connectivity!
On March 16, 2015, 1:57 p.m., Matt Jordan wrote: /branches/13/configs/basic-pbx/extensions.conf, lines 36-42 https://reviewboard.asterisk.org/r/4488/diff/1/?file=72117#file72117line36 Does this really need to be a separate context? I'm all for having contexts break up logical groupings of subroutines, but the dialplan here feels like it is getting a bit out of control. Subroutines can already be named via an actual extension name - when you have 'catch alls' in the various contexts, that feels like a sign that things aren't being set up at the right granularity. For example, this could just be in your [Internal] context: [Internal] exten = internal_setup,1,NoOp() same = n,Set(CDR_PROP(disable)=1) same = n,Return() Instead of invoking this with an explicit Goto, use GoSub. That way it can be called from anywhere, and doesn't require a lot of Gotos to jump around. Generally, you should always prefer Goto over GoSub, unless you *really* want them to leave the context they are in. Jonathan Rose wrote: Hmmm, the reason he did this was because he didn't want to add setup code to every extension in the [Internal] context. Using a gosub works here, but it will still require invoking the gosub on every extension and what he wanted to do was just automatically call this stuff on everything that goes into the Internal context. Plus this way the CDR disabling stuff gets ran when an internal context calls into an extension that is just included by Internal. Matt Jordan wrote: Hm. That's a fair point. I'd rename this slightly then: have the Pre-Internal be Internal - since it should be applied to all internal extensions - and place the actual internal extensions into some other context. rnewton wrote: I'm changing Pre-Internal to Internal and Internal to General as the contexts included there are not really specific to internal use. Hah I forgot about the default general context. That caused me some confusion. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4488/#review14698 --- On March 13, 2015, 2:32 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4488/ --- (Updated March 13, 2015, 2:32 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- Howdy, here is another patch for the Super Awesome Company configuration. We are still in phase 1. The general requirements are posted on the wiki: https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company The specific requirements this patch meets are below: pjsip.conf * SIP ITSP configuration example and have place holders for the required authentication bits. ** Assume that Asterisk does not have a public IP address, and sits behind a NAT with its desk phones. * Have outbound registration to the SIP trunk, and an endpoint that represents the SIP trunk. * Inbound calls received from the SIP trunk should go into their own context. extensions.conf * Match the outbound dial request so that it can only dial US area codes. ** Don't let people dial 900 numbers, international numbers, or any other numbers that could result in a charge * Inbound calls from the SIP trunk should hit a basic Auto Attendant that prompts them for the extension to dial, after greeting them to SAC. * If an inbound call matches a DID that maps to a specific extension/device, dial that extension/device directly. Billing * Make sure CDRs output all calls that are from/to the SIP trunk. These should be logged to a CSV. * For intra-office calls, kill the CDRs. Additional Requirements Noted: * For outbound calls, each SAC employee’s 10-digit DID number is provided as their Caller ID. * Voicemail may be accessed remotely by employees who dial 256-555-1234. When employees dial voicemail remotely, they must input both their mailbox number and their pin code. * 7, 10 and 10+1 digit dialing for local and long distance calls. * Internal dialing of otherwise inbound features, ** 1100 to reach the main IVR. * The IVR options possible without getting into Phase 2. Diffs - /branches/13/configs/basic-pbx/pjsip.conf 432866 /branches/13/configs/basic-pbx/modules.conf 432866 /branches/13/configs/basic-pbx/logger.conf 432866 /branches/13/configs/basic-pbx/extensions.conf 432866 Diff: https://reviewboard.asterisk.org/r/4488/diff/ Testing --- Setup with a Digium Cloud Services trunk and a few internal phones
Re: [asterisk-dev] [Code Review] 4488: Super Awesome Company: Phase 1 - Patch 2 - Outside Connectivity!
On March 16, 2015, 1:57 p.m., Matt Jordan wrote:
/branches/13/configs/basic-pbx/extensions.conf, lines 135-136
https://reviewboard.asterisk.org/r/4488/diff/1/?file=72117#file72117line135
I'm assuming we're going to replace the prompts eventually? :-)
rnewton wrote:
Yeah I wasn't sure if we wanted to deliver some custom sounds with this
example or just placeholders. It would be nice if we had some custom sounds
to go with it.
If we did, where would be the best place for the custom, example-specific
sounds to live in the source?
Who do we have record the sounds? A professional? Or just me?
Matt Jordan wrote:
We could always ask Allison :-)
:D Alrighty. I'll use currently available sounds as placeholders to avoid
problems. In the meantime I'll make a new issue to go ask Allison after we are
a bit farther down the line and are sure what all prompts we may need.
On March 16, 2015, 1:57 p.m., Matt Jordan wrote:
/branches/13/configs/basic-pbx/extensions.conf, lines 64-67
https://reviewboard.asterisk.org/r/4488/diff/1/?file=72117#file72117line64
If this is a subroutine, it needs to Return().
rnewton wrote:
It Returns on the other end of the Goto operations. Is that bad?
Matt Jordan wrote:
You don't want to mix idioms.
If you are using Goto, then Goto to the extension.
If you are using Gosub, then always Gosub to the extension.
Subroutines should be callable from anywhere, and should not impact the
call flow. Goto should be used when the call flow leads to logically move to
another extension, and will never return from that point.
I was using Gosub in places where I expected we would need them in the future,
but that is probably bad form. I modified things now to reflect only what is
necessary now. They will pretty much all be Goto when I update the diff.
On March 16, 2015, 1:57 p.m., Matt Jordan wrote:
/branches/13/configs/basic-pbx/extensions.conf, line 67
https://reviewboard.asterisk.org/r/4488/diff/1/?file=72117#file72117line67
Since your 'dialed-${DIALSTATUS}' extensions are subroutines, this
needs to be invoked as a subroutine.
rnewton wrote:
Ah, I thought Goto could be used within the operation of a Gosub and that
any Return would Return out of the current Gosub. I probably misunderstood
something fundamental about Gosubs.
These were not intended to be separate subroutines. I was just misunderstanding
proper usage of Gosub.
On March 16, 2015, 1:57 p.m., Matt Jordan wrote:
/branches/13/configs/basic-pbx/extensions.conf, lines 55-58
https://reviewboard.asterisk.org/r/4488/diff/1/?file=72117#file72117line55
You have a Gosub here without a Return. That will unbalance the stack.
rnewton wrote:
Where is the Return needed? On the h or o extensions? I'm still used to
macros so I'm not skilled with the ol' Gosub yet.
Matt Jordan wrote:
Whenever you invoke a subroutine using GoSub, that subroutine *must* end
with Return().
Everywhere that subroutine ended it would call Return or else Hangup. Though,
at the moment we don't really need a Gosub here so I switched to a Goto and
adjusted things accordingly.
- rnewton
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4488/#review14698
---
On March 13, 2015, 2:32 p.m., rnewton wrote:
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4488/
---
(Updated March 13, 2015, 2:32 p.m.)
Review request for Asterisk Developers.
Repository: Asterisk
Description
---
Howdy, here is another patch for the Super Awesome Company configuration. We
are still in phase 1. The general requirements are posted on the wiki:
https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company
The specific requirements this patch meets are below:
pjsip.conf
* SIP ITSP configuration example and have place holders for the required
authentication bits.
** Assume that Asterisk does not have a public IP address, and sits behind a
NAT with its desk phones.
* Have outbound registration to the SIP trunk, and an endpoint that
represents the SIP trunk.
* Inbound calls received from the SIP trunk should go into their own context.
extensions.conf
* Match the outbound dial request so that it can only dial US area codes.
** Don't let people dial 900 numbers, international numbers, or any other
numbers that could result in a charge
* Inbound calls from the SIP trunk should hit a basic Auto Attendant that
prompts them for the extension to dial, after greeting them to SAC.
* If an inbound call matches a DID that maps to a specific
Re: [asterisk-dev] [Code Review] 4488: Super Awesome Company: Phase 1 - Patch 2 - Outside Connectivity!
On March 16, 2015, 1:57 p.m., Matt Jordan wrote:
/branches/13/configs/basic-pbx/extensions.conf, lines 55-58
https://reviewboard.asterisk.org/r/4488/diff/1/?file=72117#file72117line55
You have a Gosub here without a Return. That will unbalance the stack.
Where is the Return needed? On the h or o extensions? I'm still used to macros
so I'm not skilled with the ol' Gosub yet.
On March 16, 2015, 1:57 p.m., Matt Jordan wrote:
/branches/13/configs/basic-pbx/extensions.conf, lines 64-67
https://reviewboard.asterisk.org/r/4488/diff/1/?file=72117#file72117line64
If this is a subroutine, it needs to Return().
It Returns on the other end of the Goto operations. Is that bad?
On March 16, 2015, 1:57 p.m., Matt Jordan wrote:
/branches/13/configs/basic-pbx/extensions.conf, line 67
https://reviewboard.asterisk.org/r/4488/diff/1/?file=72117#file72117line67
Since your 'dialed-${DIALSTATUS}' extensions are subroutines, this
needs to be invoked as a subroutine.
Ah, I thought Goto could be used within the operation of a Gosub and that any
Return would Return out of the current Gosub. I probably misunderstood
something fundamental about Gosubs.
- rnewton
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4488/#review14698
---
On March 13, 2015, 2:32 p.m., rnewton wrote:
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4488/
---
(Updated March 13, 2015, 2:32 p.m.)
Review request for Asterisk Developers.
Repository: Asterisk
Description
---
Howdy, here is another patch for the Super Awesome Company configuration. We
are still in phase 1. The general requirements are posted on the wiki:
https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company
The specific requirements this patch meets are below:
pjsip.conf
* SIP ITSP configuration example and have place holders for the required
authentication bits.
** Assume that Asterisk does not have a public IP address, and sits behind a
NAT with its desk phones.
* Have outbound registration to the SIP trunk, and an endpoint that
represents the SIP trunk.
* Inbound calls received from the SIP trunk should go into their own context.
extensions.conf
* Match the outbound dial request so that it can only dial US area codes.
** Don't let people dial 900 numbers, international numbers, or any other
numbers that could result in a charge
* Inbound calls from the SIP trunk should hit a basic Auto Attendant that
prompts them for the extension to dial, after greeting them to SAC.
* If an inbound call matches a DID that maps to a specific extension/device,
dial that extension/device directly.
Billing
* Make sure CDRs output all calls that are from/to the SIP trunk. These
should be logged to a CSV.
* For intra-office calls, kill the CDRs.
Additional Requirements Noted:
* For outbound calls, each SAC employee’s 10-digit DID number is provided as
their Caller ID.
* Voicemail may be accessed remotely by employees who dial 256-555-1234.
When employees dial voicemail remotely, they must input both their mailbox
number and their pin code.
* 7, 10 and 10+1 digit dialing for local and long distance calls.
* Internal dialing of otherwise inbound features,
** 1100 to reach the main IVR.
* The IVR options possible without getting into Phase 2.
Diffs
-
/branches/13/configs/basic-pbx/pjsip.conf 432866
/branches/13/configs/basic-pbx/modules.conf 432866
/branches/13/configs/basic-pbx/logger.conf 432866
/branches/13/configs/basic-pbx/extensions.conf 432866
Diff: https://reviewboard.asterisk.org/r/4488/diff/
Testing
---
Setup with a Digium Cloud Services trunk and a few internal phones.
Internal to Internal calls.
Calls Internal to voicemail and other features.
External to internal DID calls.
External to internal feature calls.
Basically tried to call as many ways as I could through all the various
features. Everything seemed to work.
Thanks,
rnewton
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-dev] [Code Review] 4488: Super Awesome Company: Phase 1 - Patch 2 - Outside Connectivity!
On March 16, 2015, 1:57 p.m., Matt Jordan wrote: /branches/13/configs/basic-pbx/extensions.conf, lines 36-42 https://reviewboard.asterisk.org/r/4488/diff/1/?file=72117#file72117line36 Does this really need to be a separate context? I'm all for having contexts break up logical groupings of subroutines, but the dialplan here feels like it is getting a bit out of control. Subroutines can already be named via an actual extension name - when you have 'catch alls' in the various contexts, that feels like a sign that things aren't being set up at the right granularity. For example, this could just be in your [Internal] context: [Internal] exten = internal_setup,1,NoOp() same = n,Set(CDR_PROP(disable)=1) same = n,Return() Instead of invoking this with an explicit Goto, use GoSub. That way it can be called from anywhere, and doesn't require a lot of Gotos to jump around. Generally, you should always prefer Goto over GoSub, unless you *really* want them to leave the context they are in. Jonathan Rose wrote: Hmmm, the reason he did this was because he didn't want to add setup code to every extension in the [Internal] context. Using a gosub works here, but it will still require invoking the gosub on every extension and what he wanted to do was just automatically call this stuff on everything that goes into the Internal context. Plus this way the CDR disabling stuff gets ran when an internal context calls into an extension that is just included by Internal. Matt Jordan wrote: Hm. That's a fair point. I'd rename this slightly then: have the Pre-Internal be Internal - since it should be applied to all internal extensions - and place the actual internal extensions into some other context. I'm changing Pre-Internal to Internal and Internal to General as the contexts included there are not really specific to internal use. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4488/#review14698 --- On March 13, 2015, 2:32 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4488/ --- (Updated March 13, 2015, 2:32 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- Howdy, here is another patch for the Super Awesome Company configuration. We are still in phase 1. The general requirements are posted on the wiki: https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company The specific requirements this patch meets are below: pjsip.conf * SIP ITSP configuration example and have place holders for the required authentication bits. ** Assume that Asterisk does not have a public IP address, and sits behind a NAT with its desk phones. * Have outbound registration to the SIP trunk, and an endpoint that represents the SIP trunk. * Inbound calls received from the SIP trunk should go into their own context. extensions.conf * Match the outbound dial request so that it can only dial US area codes. ** Don't let people dial 900 numbers, international numbers, or any other numbers that could result in a charge * Inbound calls from the SIP trunk should hit a basic Auto Attendant that prompts them for the extension to dial, after greeting them to SAC. * If an inbound call matches a DID that maps to a specific extension/device, dial that extension/device directly. Billing * Make sure CDRs output all calls that are from/to the SIP trunk. These should be logged to a CSV. * For intra-office calls, kill the CDRs. Additional Requirements Noted: * For outbound calls, each SAC employee’s 10-digit DID number is provided as their Caller ID. * Voicemail may be accessed remotely by employees who dial 256-555-1234. When employees dial voicemail remotely, they must input both their mailbox number and their pin code. * 7, 10 and 10+1 digit dialing for local and long distance calls. * Internal dialing of otherwise inbound features, ** 1100 to reach the main IVR. * The IVR options possible without getting into Phase 2. Diffs - /branches/13/configs/basic-pbx/pjsip.conf 432866 /branches/13/configs/basic-pbx/modules.conf 432866 /branches/13/configs/basic-pbx/logger.conf 432866 /branches/13/configs/basic-pbx/extensions.conf 432866 Diff: https://reviewboard.asterisk.org/r/4488/diff/ Testing --- Setup with a Digium Cloud Services trunk and a few internal phones. Internal to Internal calls. Calls Internal to voicemail and other features. External to internal
Re: [asterisk-dev] [Code Review] 4488: Super Awesome Company: Phase 1 - Patch 2 - Outside Connectivity!
On March 16, 2015, 1:57 p.m., Matt Jordan wrote: /branches/13/configs/basic-pbx/extensions.conf, lines 135-136 https://reviewboard.asterisk.org/r/4488/diff/1/?file=72117#file72117line135 I'm assuming we're going to replace the prompts eventually? :-) Yeah I wasn't sure if we wanted to deliver some custom sounds with this example or just placeholders. It would be nice if we had some custom sounds to go with it. If we did, where would be the best place for the custom, example-specific sounds to live in the source? Who do we have record the sounds? A professional? Or just me? On March 16, 2015, 1:57 p.m., Matt Jordan wrote: /branches/13/configs/basic-pbx/modules.conf, line 25 https://reviewboard.asterisk.org/r/4488/diff/1/?file=72119#file72119line25 Don't you need a config file for this? Forgot to 'svn add' cdr.conf and cdr_custom.conf. Whoops. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4488/#review14698 --- On March 13, 2015, 2:32 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4488/ --- (Updated March 13, 2015, 2:32 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- Howdy, here is another patch for the Super Awesome Company configuration. We are still in phase 1. The general requirements are posted on the wiki: https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company The specific requirements this patch meets are below: pjsip.conf * SIP ITSP configuration example and have place holders for the required authentication bits. ** Assume that Asterisk does not have a public IP address, and sits behind a NAT with its desk phones. * Have outbound registration to the SIP trunk, and an endpoint that represents the SIP trunk. * Inbound calls received from the SIP trunk should go into their own context. extensions.conf * Match the outbound dial request so that it can only dial US area codes. ** Don't let people dial 900 numbers, international numbers, or any other numbers that could result in a charge * Inbound calls from the SIP trunk should hit a basic Auto Attendant that prompts them for the extension to dial, after greeting them to SAC. * If an inbound call matches a DID that maps to a specific extension/device, dial that extension/device directly. Billing * Make sure CDRs output all calls that are from/to the SIP trunk. These should be logged to a CSV. * For intra-office calls, kill the CDRs. Additional Requirements Noted: * For outbound calls, each SAC employee’s 10-digit DID number is provided as their Caller ID. * Voicemail may be accessed remotely by employees who dial 256-555-1234. When employees dial voicemail remotely, they must input both their mailbox number and their pin code. * 7, 10 and 10+1 digit dialing for local and long distance calls. * Internal dialing of otherwise inbound features, ** 1100 to reach the main IVR. * The IVR options possible without getting into Phase 2. Diffs - /branches/13/configs/basic-pbx/pjsip.conf 432866 /branches/13/configs/basic-pbx/modules.conf 432866 /branches/13/configs/basic-pbx/logger.conf 432866 /branches/13/configs/basic-pbx/extensions.conf 432866 Diff: https://reviewboard.asterisk.org/r/4488/diff/ Testing --- Setup with a Digium Cloud Services trunk and a few internal phones. Internal to Internal calls. Calls Internal to voicemail and other features. External to internal DID calls. External to internal feature calls. Basically tried to call as many ways as I could through all the various features. Everything seemed to work. Thanks, rnewton -- _ -- 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] 4488: Super Awesome Company: Phase 1 - Patch 2 - Outside Connectivity!
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4488/ --- Review request for Asterisk Developers. Repository: Asterisk Description --- Howdy, here is another patch for the Super Awesome Company configuration. We are still in phase 1. The general requirements are posted on the wiki: https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company The specific requirements this patch meets are below: pjsip.conf * SIP ITSP configuration example and have place holders for the required authentication bits. ** Assume that Asterisk does not have a public IP address, and sits behind a NAT with its desk phones. * Have outbound registration to the SIP trunk, and an endpoint that represents the SIP trunk. * Inbound calls received from the SIP trunk should go into their own context. extensions.conf * Match the outbound dial request so that it can only dial US area codes. ** Don't let people dial 900 numbers, international numbers, or any other numbers that could result in a charge * Inbound calls from the SIP trunk should hit a basic Auto Attendant that prompts them for the extension to dial, after greeting them to SAC. * If an inbound call matches a DID that maps to a specific extension/device, dial that extension/device directly. Billing * Make sure CDRs output all calls that are from/to the SIP trunk. These should be logged to a CSV. * For intra-office calls, kill the CDRs. Additional Requirements Noted: * For outbound calls, each SAC employee’s 10-digit DID number is provided as their Caller ID. * Voicemail may be accessed remotely by employees who dial 256-555-1234. When employees dial voicemail remotely, they must input both their mailbox number and their pin code. * 7, 10 and 10+1 digit dialing for local and long distance calls. * Internal dialing of otherwise inbound features, ** 1100 to reach the main IVR. * The IVR options possible without getting into Phase 2. Diffs - /branches/13/configs/basic-pbx/pjsip.conf 432866 /branches/13/configs/basic-pbx/modules.conf 432866 /branches/13/configs/basic-pbx/logger.conf 432866 /branches/13/configs/basic-pbx/extensions.conf 432866 Diff: https://reviewboard.asterisk.org/r/4488/diff/ Testing --- Setup with a Digium Cloud Services trunk and a few internal phones. Internal to Internal calls. Calls Internal to voicemail and other features. External to internal DID calls. External to internal feature calls. Basically tried to call as many ways as I could through all the various features. Everything seemed to work. Thanks, rnewton -- _ -- 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] 4488: Super Awesome Company: Phase 1 - Patch 2 - Outside Connectivity!
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4488/#review14676 --- /branches/13/configs/basic-pbx/extensions.conf https://reviewboard.asterisk.org/r/4488/#comment25236 Blob, oops. /branches/13/configs/basic-pbx/logger.conf https://reviewboard.asterisk.org/r/4488/#comment25237 This snuck in there during troubleshooting apparently. I should comment that back out. - rnewton On March 13, 2015, 2:32 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4488/ --- (Updated March 13, 2015, 2:32 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- Howdy, here is another patch for the Super Awesome Company configuration. We are still in phase 1. The general requirements are posted on the wiki: https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company The specific requirements this patch meets are below: pjsip.conf * SIP ITSP configuration example and have place holders for the required authentication bits. ** Assume that Asterisk does not have a public IP address, and sits behind a NAT with its desk phones. * Have outbound registration to the SIP trunk, and an endpoint that represents the SIP trunk. * Inbound calls received from the SIP trunk should go into their own context. extensions.conf * Match the outbound dial request so that it can only dial US area codes. ** Don't let people dial 900 numbers, international numbers, or any other numbers that could result in a charge * Inbound calls from the SIP trunk should hit a basic Auto Attendant that prompts them for the extension to dial, after greeting them to SAC. * If an inbound call matches a DID that maps to a specific extension/device, dial that extension/device directly. Billing * Make sure CDRs output all calls that are from/to the SIP trunk. These should be logged to a CSV. * For intra-office calls, kill the CDRs. Additional Requirements Noted: * For outbound calls, each SAC employee’s 10-digit DID number is provided as their Caller ID. * Voicemail may be accessed remotely by employees who dial 256-555-1234. When employees dial voicemail remotely, they must input both their mailbox number and their pin code. * 7, 10 and 10+1 digit dialing for local and long distance calls. * Internal dialing of otherwise inbound features, ** 1100 to reach the main IVR. * The IVR options possible without getting into Phase 2. Diffs - /branches/13/configs/basic-pbx/pjsip.conf 432866 /branches/13/configs/basic-pbx/modules.conf 432866 /branches/13/configs/basic-pbx/logger.conf 432866 /branches/13/configs/basic-pbx/extensions.conf 432866 Diff: https://reviewboard.asterisk.org/r/4488/diff/ Testing --- Setup with a Digium Cloud Services trunk and a few internal phones. Internal to Internal calls. Calls Internal to voicemail and other features. External to internal DID calls. External to internal feature calls. Basically tried to call as many ways as I could through all the various features. Everything seemed to work. Thanks, rnewton -- _ -- 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] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Feb. 25, 2015, 5:48 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 432301 Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible. voicemail.conf * Set up mailboxes for every person in SAC. Assign 'default' pins. Create reasonable basic settings. * Do not set up e-mail or pager addresses. REVIEW? Please, if possible look at this from a few angles: * Use the configuration, configure a couple phones and call between them. Leave voicemails and retrieve them. * Have I created any security issues? * Is my dialplan easy to understand? * Could anything be done more efficiently without making it over-complicated? * Have I over-complicated anything? * Are there any critical settings I'm missing from any of the files? A couple, more specific questions: * We have sample configs in /configs/samples; what directory do we want these configurations in? (I used /configs/examples for now, but I don't really like it) * We have the make target make samples for the current samples; what do we want for these new configs? Diffs - /branches/13/configs/examples/awesome/voicemail.conf PRE-CREATION /branches/13/configs/examples/awesome/pjsip.conf PRE-CREATION /branches/13/configs/examples/awesome/musiconhold.conf PRE-CREATION /branches/13/configs/examples/awesome/modules.conf PRE-CREATION /branches/13/configs/examples/awesome/logger.conf PRE-CREATION /branches/13/configs/examples/awesome/indications.conf PRE-CREATION /branches/13/configs/examples/awesome/extensions.conf PRE-CREATION /branches/13/configs/examples/awesome/asterisk.conf PRE-CREATION /branches/13/configs/examples/awesome/README PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4379/diff/ Testing --- Setup Asterisk with configuration, connected up three phones using the first three users. Made calls between them all, left voicemails and retrieved them with all users. Verified MWI working with all phones. Thanks, rnewton -- _ -- 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] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
On Feb. 20, 2015, 2:37 p.m., Matt Jordan wrote: /branches/13/configs/examples/awesome/extensions.conf, lines 29-31 https://reviewboard.asterisk.org/r/4379/diff/2/?file=71364#file71364line29 Is this comment correct still with this pattern match? Nope. Fixed. I also adjusted the pattern match here and for hints as for the current users we only need '_11XX' . - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14511 --- On Feb. 13, 2015, 12:46 a.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Feb. 13, 2015, 12:46 a.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible. voicemail.conf * Set up mailboxes for every person in SAC. Assign 'default' pins. Create reasonable basic settings. * Do not set up e-mail or pager addresses. REVIEW? Please, if possible look at this from a few angles: * Use the configuration, configure a couple phones and call between them. Leave voicemails and retrieve them. * Have I created any security issues? * Is my dialplan easy to understand? * Could anything be done more efficiently without making it over-complicated? * Have I over-complicated anything? * Are there any critical settings I'm missing from any of the files? A couple, more specific questions: * We have sample configs in /configs/samples; what directory do we want these configurations in? (I used /configs/examples for now, but I don't really like it) * We have the make target make samples for the current samples; what do we want for these new configs? Diffs - /branches/13/configs/examples/awesome/voicemail.conf PRE-CREATION /branches/13/configs/examples/awesome/pjsip.conf PRE-CREATION /branches/13/configs/examples/awesome/musiconhold.conf PRE-CREATION /branches/13/configs/examples/awesome/modules.conf PRE-CREATION /branches/13/configs/examples/awesome/logger.conf PRE-CREATION /branches/13/configs/examples/awesome/indications.conf PRE-CREATION /branches/13/configs/examples/awesome/extensions.conf PRE-CREATION /branches/13/configs/examples/awesome/asterisk.conf PRE-CREATION /branches/13/configs/examples/awesome/README PRE-CREATION Diff: https
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Feb. 24, 2015, 1:20 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible. voicemail.conf * Set up mailboxes for every person in SAC. Assign 'default' pins. Create reasonable basic settings. * Do not set up e-mail or pager addresses. REVIEW? Please, if possible look at this from a few angles: * Use the configuration, configure a couple phones and call between them. Leave voicemails and retrieve them. * Have I created any security issues? * Is my dialplan easy to understand? * Could anything be done more efficiently without making it over-complicated? * Have I over-complicated anything? * Are there any critical settings I'm missing from any of the files? A couple, more specific questions: * We have sample configs in /configs/samples; what directory do we want these configurations in? (I used /configs/examples for now, but I don't really like it) * We have the make target make samples for the current samples; what do we want for these new configs? Diffs (updated) - /branches/13/configs/examples/awesome/voicemail.conf PRE-CREATION /branches/13/configs/examples/awesome/pjsip.conf PRE-CREATION /branches/13/configs/examples/awesome/musiconhold.conf PRE-CREATION /branches/13/configs/examples/awesome/modules.conf PRE-CREATION /branches/13/configs/examples/awesome/logger.conf PRE-CREATION /branches/13/configs/examples/awesome/indications.conf PRE-CREATION /branches/13/configs/examples/awesome/extensions.conf PRE-CREATION /branches/13/configs/examples/awesome/asterisk.conf PRE-CREATION /branches/13/configs/examples/awesome/README PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4379/diff/ Testing --- Setup Asterisk with configuration, connected up three phones using the first three users. Made calls between them all, left voicemails and retrieved them with all users. Verified MWI working with all phones. Thanks, rnewton -- _ -- 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] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
On Feb. 16, 2015, 10:59 p.m., Matt Jordan wrote: /branches/13/configs/examples/awesome/logger.conf, line 5 https://reviewboard.asterisk.org/r/4379/diff/2/?file=71366#file71366line5 Go with least to greatest: console = verbose,notice,warning,error That works. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14470 --- On Feb. 13, 2015, 12:46 a.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Feb. 13, 2015, 12:46 a.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible. voicemail.conf * Set up mailboxes for every person in SAC. Assign 'default' pins. Create reasonable basic settings. * Do not set up e-mail or pager addresses. REVIEW? Please, if possible look at this from a few angles: * Use the configuration, configure a couple phones and call between them. Leave voicemails and retrieve them. * Have I created any security issues? * Is my dialplan easy to understand? * Could anything be done more efficiently without making it over-complicated? * Have I over-complicated anything? * Are there any critical settings I'm missing from any of the files? A couple, more specific questions: * We have sample configs in /configs/samples; what directory do we want these configurations in? (I used /configs/examples for now, but I don't really like it) * We have the make target make samples for the current samples; what do we want for these new configs? Diffs - /branches/13/configs/examples/awesome/voicemail.conf PRE-CREATION /branches/13/configs/examples/awesome/pjsip.conf PRE-CREATION /branches/13/configs/examples/awesome/musiconhold.conf PRE-CREATION /branches/13/configs/examples/awesome/modules.conf PRE-CREATION /branches/13/configs/examples/awesome/logger.conf PRE-CREATION /branches/13/configs/examples/awesome/indications.conf PRE-CREATION /branches/13/configs/examples/awesome/extensions.conf PRE-CREATION /branches/13/configs/examples/awesome/asterisk.conf PRE-CREATION /branches/13/configs/examples/awesome/README PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4379/diff/ Testing --- Setup Asterisk with configuration
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14484 --- /branches/13/configs/examples/awesome/extensions.conf https://reviewboard.asterisk.org/r/4379/#comment25024 Forgot to remove this Verbose call that I was using for debugging an issue when testing. - rnewton On Feb. 13, 2015, 12:46 a.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Feb. 13, 2015, 12:46 a.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible. voicemail.conf * Set up mailboxes for every person in SAC. Assign 'default' pins. Create reasonable basic settings. * Do not set up e-mail or pager addresses. REVIEW? Please, if possible look at this from a few angles: * Use the configuration, configure a couple phones and call between them. Leave voicemails and retrieve them. * Have I created any security issues? * Is my dialplan easy to understand? * Could anything be done more efficiently without making it over-complicated? * Have I over-complicated anything? * Are there any critical settings I'm missing from any of the files? A couple, more specific questions: * We have sample configs in /configs/samples; what directory do we want these configurations in? (I used /configs/examples for now, but I don't really like it) * We have the make target make samples for the current samples; what do we want for these new configs? Diffs - /branches/13/configs/examples/awesome/voicemail.conf PRE-CREATION /branches/13/configs/examples/awesome/pjsip.conf PRE-CREATION /branches/13/configs/examples/awesome/musiconhold.conf PRE-CREATION /branches/13/configs/examples/awesome/modules.conf PRE-CREATION /branches/13/configs/examples/awesome/logger.conf PRE-CREATION /branches/13/configs/examples/awesome/indications.conf PRE-CREATION /branches/13/configs/examples/awesome/extensions.conf PRE-CREATION /branches/13/configs/examples/awesome/asterisk.conf PRE-CREATION /branches/13/configs/examples/awesome/README PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4379/diff/ Testing --- Setup Asterisk with configuration, connected up three phones using the first three users. Made calls between them all
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
On Feb. 16, 2015, 10:59 p.m., Matt Jordan wrote: /branches/13/configs/examples/awesome/asterisk.conf, line 1 https://reviewboard.asterisk.org/r/4379/diff/2/?file=71363#file71363line1 This isn't a template Yup. Oddly it is this way in the asterisk.conf.sample as well, and I'm not sure why. On Feb. 16, 2015, 10:59 p.m., Matt Jordan wrote: /branches/13/configs/examples/awesome/asterisk.conf, line 2 https://reviewboard.asterisk.org/r/4379/diff/2/?file=71363#file71363line2 So, this is where things get fun. In extensions.conf, we've noted that under the hood, there is no semantic difference between '=' and '=', even if previously, the nomenclature was to use '='. The same is true here, as well as in modules.conf and elsewhere. I'd prefer us to be consistent: either we should continue to use the existing nomenclature, where non-key/value pair settings use '=' (objects, as they used to be called) - or we should use '=' everywhere. Replaced all '=' with '='. I also made the spacing around '=' uniform across the files. Most settings used 'option = setting' rather than 'option=setting', so I changed them all to the former which also made things more readable. On Feb. 16, 2015, 10:59 p.m., Matt Jordan wrote: /branches/13/configs/examples/awesome/extensions.conf, lines 40-44 https://reviewboard.asterisk.org/r/4379/diff/2/?file=71364#file71364line40 I think the lining scheme in the extensions.conf is a little odd. In some places, you clearly are lining things up based on the priority. Here, however, we're not - and we're a little off of the '=' sign as well. Personally, I prefer lining things up based on the '=' or '=', as that is consistent throughout the file, and the first line in any extension should almost always be an 'entry point'. That keeps things lined up: exten = my_long_extension,1,NoOp() same = n,foo() same = n,bar() exten = a_short_one,1,NoOp() same = n,invoke_something() same = n,do_something_else() That being said, I'm also fine if we line things up based on priority - but I'd go for a consistent scheme and stick with it. (I do think lining things up on priority leads to a lot of leading white space, but that's just my laziness kicking in.) I prefer lining up on the priority, but I'm usually writing very short dialplans for testing issues. I played with both and I agree with you here. I think lining up on the '=' will be better as this dialplan grows. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14470 --- On Feb. 13, 2015, 12:46 a.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Feb. 13, 2015, 12:46 a.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
On Jan. 29, 2015, 10:23 p.m., Mark Michelson wrote: /branches/13/configs/examples/super_awesome_company/pjsip.conf, line 41 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71114#file71114line41 I'm curious why you elected to use MAC addresses as the endpoint names. I'd personally find things a lot easier to configure/maintain if the SIP endpoint/aor/auth name is the same as the voicemail box number is the same as the extension number, etc. This also means that if Lindsey does some crazy extreme stunt that smashes her phone, then when she replaces it with a new one, you're going to have to change config values everywhere to have the new MAC address of the phone. Matt Jordan wrote: Hm. I think that's usually one of those best practices. You generally don't want the auth user to be easily guessed. Of course, we could split the concept of the endpoint name from the auth user, which would then allow the endpoints to be named 107 (for example) and the auth user to be her MAC address. Joshua Colp wrote: I think in practice this would just cause problems. Not all devices allow those two things to be separate. It's annoying. Mark Michelson wrote: SAC uses Digium phones, and Digium phones allow separate user and authuser to be specified. Joshua Colp wrote: Your statement is true but it would be nice if we could err on the side of not falling into a trap of doing fundamental stuff which isn't applicable to the wide world. rnewton wrote: I used MAC addresses as that is what we use as an example in our security best practices document: http://svnview.digium.com/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt?view=markup Perhaps this is a moot point. SAC's Asterisk system is behind NAT and firewall, so we could change the spec to specify that IT has locked down traffic between Asterisk and the public internet to only allow inbound traffic from the ITSP addresses. Or, on Asterisk we can use ACL's to limit traffic allowed to the internal network and ITSP addresses. With either of those approaches we should be able to use the less secure extension numbered auth users. What would be the issues either of these approaches other than an attacker on the internal network? Matt Jordan wrote: I think we need to come to some concurrence here so that the diffs can get updated. I suspect there are going to be additional rounds of review. The purpose of this set of example configs is to provide a base for a recommended deployment. Regardless of the scheme chosen, the example absolutely should use the best practices so that people have a secure system. If someone wants to use 'alice' and 'bob' for their names, that may be suitable for some examples, but not suitable for a recommended deployment. I don't care if we use MAC address or something else that is suitably difficult to guess, but MAC address is what A:TDG recommends [1] as well as our README-SERIOUSLY [2], and that feels like a decent starting point. [1] http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id216341.html#DeviceConfig_id291081 [2] http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt Looks like the best compromise is splitting out the user ID and auth name. Therefore the endpoint,aor,auth objects will be the extension number and the auth usernames will be the MAC address. We still have security, things are easier to administrate and the only downside is the config won't work as well for some phones. That being said, even some cheap generic phones I had laying around were able to configure a separate user id/account and auth name. This also makes other parts of the dialplan easier, hints and no crazy variable mapping for the endpoint names. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14377 --- On Jan. 27, 2015, 7:15 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Jan. 27, 2015, 7:15 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Feb. 13, 2015, 12:46 a.m.) Review request for Asterisk Developers. Changes --- Basically, review it again from head to toe. Re-tested, same testing as described in initial post. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible. voicemail.conf * Set up mailboxes for every person in SAC. Assign 'default' pins. Create reasonable basic settings. * Do not set up e-mail or pager addresses. REVIEW? Please, if possible look at this from a few angles: * Use the configuration, configure a couple phones and call between them. Leave voicemails and retrieve them. * Have I created any security issues? * Is my dialplan easy to understand? * Could anything be done more efficiently without making it over-complicated? * Have I over-complicated anything? * Are there any critical settings I'm missing from any of the files? A couple, more specific questions: * We have sample configs in /configs/samples; what directory do we want these configurations in? (I used /configs/examples for now, but I don't really like it) * We have the make target make samples for the current samples; what do we want for these new configs? Diffs (updated) - /branches/13/configs/examples/awesome/voicemail.conf PRE-CREATION /branches/13/configs/examples/awesome/pjsip.conf PRE-CREATION /branches/13/configs/examples/awesome/musiconhold.conf PRE-CREATION /branches/13/configs/examples/awesome/modules.conf PRE-CREATION /branches/13/configs/examples/awesome/logger.conf PRE-CREATION /branches/13/configs/examples/awesome/indications.conf PRE-CREATION /branches/13/configs/examples/awesome/extensions.conf PRE-CREATION /branches/13/configs/examples/awesome/asterisk.conf PRE-CREATION /branches/13/configs/examples/awesome/README PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4379/diff/ Testing --- Setup Asterisk with configuration, connected up three phones using the first three users. Made calls between them all, left voicemails and retrieved them with all users. Verified MWI working with all phones. Thanks, rnewton -- _ -- 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] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14456 --- /branches/13/configs/examples/awesome/extensions.conf https://reviewboard.asterisk.org/r/4379/#comment24995 Re-phrase this and remove trailing whitespace. - rnewton On Feb. 13, 2015, 12:46 a.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Feb. 13, 2015, 12:46 a.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible. voicemail.conf * Set up mailboxes for every person in SAC. Assign 'default' pins. Create reasonable basic settings. * Do not set up e-mail or pager addresses. REVIEW? Please, if possible look at this from a few angles: * Use the configuration, configure a couple phones and call between them. Leave voicemails and retrieve them. * Have I created any security issues? * Is my dialplan easy to understand? * Could anything be done more efficiently without making it over-complicated? * Have I over-complicated anything? * Are there any critical settings I'm missing from any of the files? A couple, more specific questions: * We have sample configs in /configs/samples; what directory do we want these configurations in? (I used /configs/examples for now, but I don't really like it) * We have the make target make samples for the current samples; what do we want for these new configs? Diffs - /branches/13/configs/examples/awesome/voicemail.conf PRE-CREATION /branches/13/configs/examples/awesome/pjsip.conf PRE-CREATION /branches/13/configs/examples/awesome/musiconhold.conf PRE-CREATION /branches/13/configs/examples/awesome/modules.conf PRE-CREATION /branches/13/configs/examples/awesome/logger.conf PRE-CREATION /branches/13/configs/examples/awesome/indications.conf PRE-CREATION /branches/13/configs/examples/awesome/extensions.conf PRE-CREATION /branches/13/configs/examples/awesome/asterisk.conf PRE-CREATION /branches/13/configs/examples/awesome/README PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4379/diff/ Testing --- Setup Asterisk with configuration, connected up three phones using the first three users. Made calls between them all, left voicemails and retrieved them
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
On Jan. 27, 2015, 8:34 p.m., Matt Jordan wrote: /branches/13/configs/examples/super_awesome_company/extensions.conf, line 42 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71109#file71109line42 I know '=' is the same as '=', but I (for some reason) still prefer '=' in dialplan. I'm not sure why. Joshua Colp wrote: I'm the same way. Across the dialplans I've seen they have primarily used '='. If this uses '=' I'd actually expect people to not know it's the same and question it. rnewton wrote: I've seen both, '=' may be more common, however if they are the same then I'd like to go with '=' to encourage future users to use '='. We should start doing this through all of our configs and examples since the extra character is unnecessary. If users question it, documentation exists on the wiki stating that they are the same. That being said, if there is a lot of hate for '=' then I'm fine with '='. This is a trivial issue that I don't care a lot about. Matt Jordan wrote: I think there's a general consensus that, despite the semantics being identical, '=' is preferred for extensions and hints in the dialplan. George Joseph wrote: Why propagate something that does absolutely nothing? It's only going to cause someone new to Asterisk to ask Why is it = in extensions.conf and = everywhere else?. That was my first question when looking at the Asterisk config files for the first time. I agree with you George. We have already had issues filed about the confusion: https://issues.asterisk.org/jira/browse/ASTERISK-23032 There may only be one, or no config files that require '='. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14320 --- On Jan. 27, 2015, 7:15 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Jan. 27, 2015, 7:15 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible. voicemail.conf * Set up mailboxes for every person in SAC. Assign 'default' pins. Create reasonable basic settings. * Do not set up e-mail or pager addresses. REVIEW? Please, if possible look at this from a few angles: * Use
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
On Jan. 30, 2015, 7:51 p.m., Paul Belanger wrote: /branches/13/configs/examples/super_awesome_company/extensions.conf, line 78 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71109#file71109line78 I prefer putting include statements at the top of the context. Also, include = Yeah that does seem to make them a little easier to find when visually scanning. On Jan. 30, 2015, 7:51 p.m., Paul Belanger wrote: /branches/13/configs/examples/super_awesome_company/indications.conf, line 1 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71110#file71110line1 Again, this file seem to be the default. I'd rather see a symlink. For the sake of avoiding confusion I'll keep it as an actual file. Also, this example set is not intended to be laid over the other samples, so there will be no file to symlink to. On Jan. 30, 2015, 7:51 p.m., Paul Belanger wrote: /branches/13/configs/examples/super_awesome_company/logger.conf, line 1 https://reviewboard.asterisk.org/r/4379/diff/1/?file=7#file7line1 Seems to be the default file, rather see a symlink It isn't the default file. For the sake of avoiding confusion I'll keep it as an actual file. Also, this example set is not intended to be laid over the other samples, so there will be no file to symlink to. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14401 --- On Jan. 27, 2015, 7:15 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Jan. 27, 2015, 7:15 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible. voicemail.conf * Set up mailboxes for every person in SAC. Assign 'default' pins. Create reasonable basic settings. * Do not set up e-mail or pager addresses. REVIEW? Please, if possible look at this from a few angles: * Use the configuration, configure a couple phones and call between them. Leave voicemails and retrieve them. * Have I created any security issues? * Is my dialplan easy to understand? * Could anything be done more efficiently without making it over-complicated? * Have I over-complicated anything? * Are there any critical settings I'm missing from any of the files? A couple, more
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
On Jan. 30, 2015, 7:51 p.m., Paul Belanger wrote: /branches/13/configs/examples/super_awesome_company/indications.conf, line 1 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71110#file71110line1 Again, this file seem to be the default. I'd rather see a symlink. rnewton wrote: For the sake of avoiding confusion I'll keep it as an actual file. Also, this example set is not intended to be laid over the other samples, so there will be no file to symlink to. I forgot I did change this as well, we removed all the unnecessary countries, since this configuration assumes the company is in Waldo, Alabama, USA. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14401 --- On Jan. 27, 2015, 7:15 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Jan. 27, 2015, 7:15 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible. voicemail.conf * Set up mailboxes for every person in SAC. Assign 'default' pins. Create reasonable basic settings. * Do not set up e-mail or pager addresses. REVIEW? Please, if possible look at this from a few angles: * Use the configuration, configure a couple phones and call between them. Leave voicemails and retrieve them. * Have I created any security issues? * Is my dialplan easy to understand? * Could anything be done more efficiently without making it over-complicated? * Have I over-complicated anything? * Are there any critical settings I'm missing from any of the files? A couple, more specific questions: * We have sample configs in /configs/samples; what directory do we want these configurations in? (I used /configs/examples for now, but I don't really like it) * We have the make target make samples for the current samples; what do we want for these new configs? Diffs - /branches/13/configs/examples/super_awesome_company/voicemail.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/pjsip.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/musiconhold.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/modules.conf PRE-CREATION /branches/13/configs/examples
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
On Jan. 29, 2015, 10:23 p.m., Mark Michelson wrote: /branches/13/configs/examples/super_awesome_company/pjsip.conf, line 41 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71114#file71114line41 I'm curious why you elected to use MAC addresses as the endpoint names. I'd personally find things a lot easier to configure/maintain if the SIP endpoint/aor/auth name is the same as the voicemail box number is the same as the extension number, etc. This also means that if Lindsey does some crazy extreme stunt that smashes her phone, then when she replaces it with a new one, you're going to have to change config values everywhere to have the new MAC address of the phone. Matt Jordan wrote: Hm. I think that's usually one of those best practices. You generally don't want the auth user to be easily guessed. Of course, we could split the concept of the endpoint name from the auth user, which would then allow the endpoints to be named 107 (for example) and the auth user to be her MAC address. Joshua Colp wrote: I think in practice this would just cause problems. Not all devices allow those two things to be separate. It's annoying. Mark Michelson wrote: SAC uses Digium phones, and Digium phones allow separate user and authuser to be specified. Joshua Colp wrote: Your statement is true but it would be nice if we could err on the side of not falling into a trap of doing fundamental stuff which isn't applicable to the wide world. I used MAC addresses as that is what we use as an example in our security best practices document: http://svnview.digium.com/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt?view=markup Perhaps this is a moot point. SAC's Asterisk system is behind NAT and firewall, so we could change the spec to specify that IT has locked down traffic between Asterisk and the public internet to only allow inbound traffic from the ITSP addresses. Or, on Asterisk we can use ACL's to limit traffic allowed to the internal network and ITSP addresses. With either of those approaches we should be able to use the less secure extension numbered auth users. What would be the issues either of these approaches other than an attacker on the internal network? - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14377 --- On Jan. 27, 2015, 7:15 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Jan. 27, 2015, 7:15 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
On Jan. 27, 2015, 8:34 p.m., Matt Jordan wrote: /branches/13/configs/examples/super_awesome_company/pjsip.conf, lines 24-30 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71114#file71114line24 I'd rename these to: auth-basic aor-basic Since they are supposed to generally accompany 'endpoint-basic' My idea was rather to have the name of each object reflect the unique configuration of that object. If I go to add new endpoints, the generic titles of basic, advanced, etc won't make it quick to remember which templates I want to use for my new endpoint. What I may do is change the endpoint object title to better reflect its usage. 'endpoint-basic' isn't helpful. Maybe endpoint-internal-d70. Let me know if that makes sense. On Jan. 27, 2015, 8:34 p.m., Matt Jordan wrote: /branches/13/configs/examples/super_awesome_company/voicemail.conf, lines 9-23 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71115#file71115line9 All of them should have VoiceMail pins SAC provides an extension, 800, that lets internal employees directly dial their voicemail box. When internal SAC employees dial their voicemail box, they’re not prompted for their mailbox number or their PIN code. Voicemail may be accessed remotely by employees who dial 256-555-1234. When employees dial voicemail remotely, they must input both their mailbox number and their pin code. Aha. I read the first line, but missed the pin code mention on the second. I just remembered the 's' option for voicemailmain as well to have it ignore the pin. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14320 --- On Jan. 27, 2015, 7:15 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Jan. 27, 2015, 7:15 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible. voicemail.conf * Set up mailboxes for every person in SAC. Assign 'default' pins. Create reasonable basic settings. * Do not set up e-mail or pager addresses. REVIEW? Please, if possible look at this from a few angles: * Use the configuration, configure a couple phones and call between them. Leave voicemails and retrieve them. * Have I
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
On Jan. 30, 2015, 1:13 p.m., Joshua Colp wrote: /branches/13/configs/examples/super_awesome_company/modules.conf, line 41 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71112#file71112line41 Add format_pcm. This covers ulaw and g722 as well. On installation it's wise to have these sounds installed as well. It was already on there. :) - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14391 --- On Jan. 27, 2015, 7:15 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Jan. 27, 2015, 7:15 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible. voicemail.conf * Set up mailboxes for every person in SAC. Assign 'default' pins. Create reasonable basic settings. * Do not set up e-mail or pager addresses. REVIEW? Please, if possible look at this from a few angles: * Use the configuration, configure a couple phones and call between them. Leave voicemails and retrieve them. * Have I created any security issues? * Is my dialplan easy to understand? * Could anything be done more efficiently without making it over-complicated? * Have I over-complicated anything? * Are there any critical settings I'm missing from any of the files? A couple, more specific questions: * We have sample configs in /configs/samples; what directory do we want these configurations in? (I used /configs/examples for now, but I don't really like it) * We have the make target make samples for the current samples; what do we want for these new configs? Diffs - /branches/13/configs/examples/super_awesome_company/voicemail.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/pjsip.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/musiconhold.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/modules.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/logger.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/indications.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/extensions.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/asterisk.conf
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
On Jan. 30, 2015, 7:51 p.m., Paul Belanger wrote: /branches/13/configs/examples/super_awesome_company/README, line 2 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71107#file71107line2 maybe shorten directory name to awesome, or sae? Or, another option would be to use Acme, Inc. I think awesome works. On Jan. 30, 2015, 7:51 p.m., Paul Belanger wrote: /branches/13/configs/examples/super_awesome_company/asterisk.conf, line 1 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71108#file71108line1 My thoughts of this file is just to symbiotic link the current asterisk.conf file. Since, you are'nt really doing anything here. If someone is using this example, there shouldn't be a current asterisk.conf file. The example asterisk.conf has trimmed documentation down to the most basic of what you might want to change for a simple setup. On Jan. 30, 2015, 7:51 p.m., Paul Belanger wrote: /branches/13/configs/examples/super_awesome_company/asterisk.conf, lines 28-36 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71108#file71108line28 This seems to be dead code, as such just remove? I think we should leave it there to emphasize the importance of it. This example is both a template and an example, so they may want to uncomment these lines. It is hard to strike a balance between removing everything they might need to uncomment and leaving everything in (ending up back with the previous .conf.sample files). Should we explain more in the comments when we would uncomment it? On Jan. 30, 2015, 7:51 p.m., Paul Belanger wrote: /branches/13/configs/examples/super_awesome_company/extensions.conf, lines 6-20 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71109#file71109line6 While this is good practices, in the real world using MAC for actual extensions is bulky and complicated. I would suggest using actually 101-115 as extension numbers but using mac for the authenticate portion of your sip.conf files. Maybe we can change this. See conversation in Mark's comment on same topic. On Jan. 30, 2015, 7:51 p.m., Paul Belanger wrote: /branches/13/configs/examples/super_awesome_company/extensions.conf, line 42 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71109#file71109line42 I'd also prefer to see the first time always a naked NoOp() exten = 8000,1,NoOp() same = n,Verbose(1, blah) this way, when jumping into new contexts, you know you'll always hit a naked noop. Can you explain the usefulness of this more? On Jan. 30, 2015, 7:51 p.m., Paul Belanger wrote: /branches/13/configs/examples/super_awesome_company/extensions.conf, line 50 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71109#file71109line50 using ._ is not recommend practices and will generate warnings in asterisk. Change to _X. Oops, thought I had _X., thanks. On Jan. 30, 2015, 7:51 p.m., Paul Belanger wrote: /branches/13/configs/examples/super_awesome_company/extensions.conf, lines 53-66 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71109#file71109line53 I disagree with this comments. Moving subroutines into there own contexts allow you to reuse code between applications. However, this example does not do this, it could be possible to create generate subroutines in asterisk samples to share across my examples. I ended up rewriting this whole section for a few reasons. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14401 --- On Jan. 27, 2015, 7:15 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Jan. 27, 2015, 7:15 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
On Jan. 27, 2015, 8:34 p.m., Matt Jordan wrote: /branches/13/configs/examples/super_awesome_company/extensions.conf, line 42 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71109#file71109line42 I know '=' is the same as '=', but I (for some reason) still prefer '=' in dialplan. I'm not sure why. Joshua Colp wrote: I'm the same way. Across the dialplans I've seen they have primarily used '='. If this uses '=' I'd actually expect people to not know it's the same and question it. I've seen both, '=' may be more common, however if they are the same then I'd like to go with '=' to encourage future users to use '='. We should start doing this through all of our configs and examples since the extra character is unnecessary. If users question it, documentation exists on the wiki stating that they are the same. That being said, if there is a lot of hate for '=' then I'm fine with '='. This is a trivial issue that I don't care a lot about. On Jan. 27, 2015, 8:34 p.m., Matt Jordan wrote: /branches/13/configs/examples/super_awesome_company/extensions.conf, lines 60-66 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71109#file71109line60 I would go ahead and play either the 'busy' or 'unavailable' prompt, based again on the DIALSTATUS. That may cause you to want to rethink whether or not invoking Voicemail should be in a separate GoSub. Yeah I ended up rewriting that whole section for several reasons. On Jan. 27, 2015, 8:34 p.m., Matt Jordan wrote: /branches/13/configs/examples/super_awesome_company/modules.conf, lines 103-108 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71112#file71112line103 While Stasis is awesome, we aren't using it yet. I'd remove it. Oops. I left these in here. I originally added them as I wasn't able to get things working without loading them (warnings/errors complained). I'll investigate what the issue was. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14320 --- On Jan. 27, 2015, 7:15 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Jan. 27, 2015, 7:15 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible
[asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible. voicemail.conf * Set up mailboxes for every person in SAC. Assign 'default' pins. Create reasonable basic settings. * Do not set up e-mail or pager addresses. REVIEW? Please, if possible look at this from a few angles: * Use the configuration, configure a couple phones and call between them. Leave voicemails and retrieve them. * Have I created any security issues? * Is my dialplan easy to understand? * Could anything be done more efficiently without making it over-complicated? * Have I over-complicated anything? * Are there any critical settings I'm missing from any of the files? A couple, more specific questions: * We have sample configs in /configs/samples; what directory do we want these configurations in? (I used /configs/examples for now, but I don't really like it) * We have the make target make samples for the current samples; what do we want for these new configs? Diffs - /branches/13/configs/examples/super_awesome_company/voicemail.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/pjsip.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/musiconhold.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/modules.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/logger.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/indications.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/extensions.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/asterisk.conf PRE-CREATION /branches/13/configs/examples/super_awesome_company/README PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4379/diff/ Testing --- Setup Asterisk with configuration, connected up three phones using the first three users. Made calls between them all, left voicemails and retrieved them with all users. Verified MWI working with all phones. Thanks, rnewton -- _ -- 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] 4328: res_pjsip: Document transport selection process
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4328/#review14156 --- This does NOT currently attempt to reuse any existing connections. A new one will always be created. This is an issue being tracked at URL. I think you forgot a URL there. - rnewton On Jan. 12, 2015, 1:33 p.m., Joshua Colp wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4328/ --- (Updated Jan. 12, 2015, 1:33 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- The transport selection process of PJSIP (and by extension some of our own logic) can be a dark dark thing. To help illuminate what happens I have created a wiki page[1] which goes through (based on the message type) the process by which a transport is chosen and how it can potentially change. Stuff to look at: 1. Is this detailed enough? 2. Can you follow it? If not, how could it be made clearer? 3. Are there additional common issues that should be covered? [1] https://wiki.asterisk.org/wiki/display/~jcolp/Transport+Selection Diffs - Diff: https://reviewboard.asterisk.org/r/4328/diff/ Testing --- I opened the wiki page. It opened. 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] 4328: res_pjsip: Document transport selection process
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4328/#review14158 --- Ship it! It looks detailed enough for me. One thing I might change would be some of the headings and the page title. You have a couple sub-headings reading PJSIP Transport Selection, whereas the others do not mention PJSIP. When you find one of our wiki pages via google, the page title shows up as the dominant text in both the search result and it is included in the URL. If I understand the page then the whole page concerns transport selection for PJSIP. I would title the page PJSIP Transport selection and not worry about stating that in the sub-headings, or else use it consistently throughout the sub-headings as well as adding PJSIP to the title. You could also link some keywords to other content on the wiki. That is always helpful in-case someone lands on this page but they are not aware of some other helpful content related to the topic. Otherwise, ship it! - rnewton On Jan. 12, 2015, 1:33 p.m., Joshua Colp wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4328/ --- (Updated Jan. 12, 2015, 1:33 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- The transport selection process of PJSIP (and by extension some of our own logic) can be a dark dark thing. To help illuminate what happens I have created a wiki page[1] which goes through (based on the message type) the process by which a transport is chosen and how it can potentially change. Stuff to look at: 1. Is this detailed enough? 2. Can you follow it? If not, how could it be made clearer? 3. Are there additional common issues that should be covered? [1] https://wiki.asterisk.org/wiki/display/~jcolp/Transport+Selection Diffs - Diff: https://reviewboard.asterisk.org/r/4328/diff/ Testing --- I opened the wiki page. It opened. 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] 4328: res_pjsip: Document transport selection process
On Jan. 12, 2015, 5:32 p.m., rnewton wrote: It looks detailed enough for me. One thing I might change would be some of the headings and the page title. You have a couple sub-headings reading PJSIP Transport Selection, whereas the others do not mention PJSIP. When you find one of our wiki pages via google, the page title shows up as the dominant text in both the search result and it is included in the URL. If I understand the page then the whole page concerns transport selection for PJSIP. I would title the page PJSIP Transport selection and not worry about stating that in the sub-headings, or else use it consistently throughout the sub-headings as well as adding PJSIP to the title. You could also link some keywords to other content on the wiki. That is always helpful in-case someone lands on this page but they are not aware of some other helpful content related to the topic. Otherwise, ship it! Oh! Add the standard Table of Contents format that we use. See the General Asterisk Wiki Page template for an example. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4328/#review14158 --- On Jan. 12, 2015, 1:33 p.m., Joshua Colp wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4328/ --- (Updated Jan. 12, 2015, 1:33 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- The transport selection process of PJSIP (and by extension some of our own logic) can be a dark dark thing. To help illuminate what happens I have created a wiki page[1] which goes through (based on the message type) the process by which a transport is chosen and how it can potentially change. Stuff to look at: 1. Is this detailed enough? 2. Can you follow it? If not, how could it be made clearer? 3. Are there additional common issues that should be covered? [1] https://wiki.asterisk.org/wiki/display/~jcolp/Transport+Selection Diffs - Diff: https://reviewboard.asterisk.org/r/4328/diff/ Testing --- I opened the wiki page. It opened. 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] 3854: manager.c - Improve documentation for manager command Getvar, Setvar
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3854/ --- (Updated Aug. 4, 2014, 2:42 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 419942 Bugs: ASTERISK-21178 https://issues.asterisk.org/jira/browse/ASTERISK-21178 Repository: Asterisk Description --- The documentation wasn't clear that AMI Getvar and Setvar could accept function calls. This is a slight modification to improve clarity. Diffs - /branches/1.8/main/manager.c 419821 Diff: https://reviewboard.asterisk.org/r/3854/diff/ Testing --- Once finalized I'll build in dev-mode with it to make sure I didn't screw up any tags. 7/30/14 @ 5:52PM CDT - Builds with no issues in dev-mode. Thanks, rnewton -- _ -- 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] 3854: manager.c - Improve documentation for manager command Getvar, Setvar
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3854/ --- (Updated Aug. 1, 2014, 6:17 p.m.) Review request for Asterisk Developers. Changes --- In regards to a conversation with M.Jordan, I removed text about variable inheritance as there is no need to be overly explicit even if the Set application help text already is. Variable inheritance is more general and is documented on the wiki already. Bugs: ASTERISK-21178 https://issues.asterisk.org/jira/browse/ASTERISK-21178 Repository: Asterisk Description --- The documentation wasn't clear that AMI Getvar and Setvar could accept function calls. This is a slight modification to improve clarity. Diffs (updated) - /branches/1.8/main/manager.c 419821 Diff: https://reviewboard.asterisk.org/r/3854/diff/ Testing --- Once finalized I'll build in dev-mode with it to make sure I didn't screw up any tags. 7/30/14 @ 5:52PM CDT - Builds with no issues in dev-mode. Thanks, rnewton -- _ -- 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] 3854: manager.c - Improve documentation for manager command Getvar, Setvar
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3854/ --- (Updated July 30, 2014, 10:40 p.m.) Review request for Asterisk Developers. Changes --- Integrated Josh and Mark's comments. Additionally, I modified the descriptions of the parameters for clarity. For example, for Setvar, the Variable parameter can really be set to a variable name, function or expression. I didn't mention expressions in the descriptions to avoid being overly verbose and explicit. Bugs: ASTERISK-21178 https://issues.asterisk.org/jira/browse/ASTERISK-21178 Repository: Asterisk Description --- The documentation wasn't clear that AMI Getvar and Setvar could accept function calls. This is a slight modification to improve clarity. Diffs (updated) - /branches/1.8/main/manager.c 419821 Diff: https://reviewboard.asterisk.org/r/3854/diff/ Testing --- Once finalized I'll build in dev-mode with it to make sure I didn't screw up any tags. Thanks, rnewton -- _ -- 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] 3854: manager.c - Improve documentation for manager command Getvar, Setvar
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3854/#review12930 --- /branches/1.8/main/manager.c https://reviewboard.asterisk.org/r/3854/#comment23315 I think I need literal tags for the mentions of underscores here and just below. - rnewton On July 30, 2014, 10:40 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3854/ --- (Updated July 30, 2014, 10:40 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-21178 https://issues.asterisk.org/jira/browse/ASTERISK-21178 Repository: Asterisk Description --- The documentation wasn't clear that AMI Getvar and Setvar could accept function calls. This is a slight modification to improve clarity. Diffs - /branches/1.8/main/manager.c 419821 Diff: https://reviewboard.asterisk.org/r/3854/diff/ Testing --- Once finalized I'll build in dev-mode with it to make sure I didn't screw up any tags. Thanks, rnewton -- _ -- 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] 3854: manager.c - Improve documentation for manager command Getvar, Setvar
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3854/ --- (Updated July 30, 2014, 10:45 p.m.) Review request for Asterisk Developers. Changes --- Add literal tags where '_' and '__' were used. Bugs: ASTERISK-21178 https://issues.asterisk.org/jira/browse/ASTERISK-21178 Repository: Asterisk Description --- The documentation wasn't clear that AMI Getvar and Setvar could accept function calls. This is a slight modification to improve clarity. Diffs (updated) - /branches/1.8/main/manager.c 419821 Diff: https://reviewboard.asterisk.org/r/3854/diff/ Testing --- Once finalized I'll build in dev-mode with it to make sure I didn't screw up any tags. Thanks, rnewton -- _ -- 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] 3854: manager.c - Improve documentation for manager command Getvar, Setvar
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3854/ --- (Updated July 30, 2014, 10:52 p.m.) Review request for Asterisk Developers. Changes --- modified testing field to describe time of test and success. Bugs: ASTERISK-21178 https://issues.asterisk.org/jira/browse/ASTERISK-21178 Repository: Asterisk Description --- The documentation wasn't clear that AMI Getvar and Setvar could accept function calls. This is a slight modification to improve clarity. Diffs - /branches/1.8/main/manager.c 419821 Diff: https://reviewboard.asterisk.org/r/3854/diff/ Testing (updated) --- Once finalized I'll build in dev-mode with it to make sure I didn't screw up any tags. 7/30/14 @ 5:52PM CDT - Builds with no issues in dev-mode. Thanks, rnewton -- _ -- 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] 3854: manager.c - Improve documentation for manager command Getvar, Setvar
On July 29, 2014, 2:32 p.m., Joshua Colp wrote: /branches/1.8/main/manager.c, line 207 https://reviewboard.asterisk.org/r/3854/diff/1/?file=65275#file65275line207 This confuses me, even after reading a few times. Specifically what you've added: directly or via function. Dialplan functions usually don't set variables. Set a channel variable or execute a dialplan function. is clearer Same applies for the other modifications. Thoughts? That makes sense to me. I believe I was thinking of using CHANNEL(blah) with Setvar as setting a channel variable via a function, but I'm betting those arguments are not considered channel variables. Could you use Setvar to just execute a dialplan function without assigning a value to the function? That is, can you really use Setvar to execute *any* dialplan function? Also, would it make sense to just use the help text description from the Set application for the description of the Setvar manager command? Albeit with a few changes: This command can be used to set the value of channel variables or dialplan functions. When setting variables, if the variable name is prefixed with '_', the variable will be inherited into channels created from the current channel. If the variable name is prefixed with '__', the variable will be inherited into channels created from the current channel and all children channels. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3854/#review12902 --- On July 25, 2014, 2:20 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3854/ --- (Updated July 25, 2014, 2:20 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-21178 https://issues.asterisk.org/jira/browse/ASTERISK-21178 Repository: Asterisk Description --- The documentation wasn't clear that AMI Getvar and Setvar could accept function calls. This is a slight modification to improve clarity. Diffs - /branches/1.8/main/manager.c 419562 Diff: https://reviewboard.asterisk.org/r/3854/diff/ Testing --- Once finalized I'll build in dev-mode with it to make sure I didn't screw up any tags. Thanks, rnewton -- _ -- 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] 3854: manager.c - Improve documentation for manager command Getvar, Setvar
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3854/ --- (Updated July 25, 2014, 2:19 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-21178 https://issues.asterisk.org/jira/browse/ASTERISK-21178 Repository: Asterisk Description --- The documentation wasn't clear that AMI Getvar and Setvar could accept function calls. This is a slight modification to improve clarity. Diffs - Diff: https://reviewboard.asterisk.org/r/3854/diff/ Testing --- Once finalized I'll build in dev-mode with it to make sure I didn't screw up any tags. Thanks, rnewton -- _ -- 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] 3854: manager.c - Improve documentation for manager command Getvar, Setvar
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3854/ --- Review request for Asterisk Developers. Bugs: ASTERISK-21178 https://issues.asterisk.org/jira/browse/ASTERISK-21178 Repository: Asterisk Description --- The documentation wasn't clear that AMI Getvar and Setvar could accept function calls. This is a slight modification to improve clarity. Diffs - Diff: https://reviewboard.asterisk.org/r/3854/diff/ Testing --- Once finalized I'll build in dev-mode with it to make sure I didn't screw up any tags. Thanks, rnewton -- _ -- 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] 3854: manager.c - Improve documentation for manager command Getvar, Setvar
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3854/ --- (Updated July 25, 2014, 2:20 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-21178 https://issues.asterisk.org/jira/browse/ASTERISK-21178 Repository: Asterisk Description --- The documentation wasn't clear that AMI Getvar and Setvar could accept function calls. This is a slight modification to improve clarity. Diffs (updated) - /branches/1.8/main/manager.c 419562 Diff: https://reviewboard.asterisk.org/r/3854/diff/ Testing --- Once finalized I'll build in dev-mode with it to make sure I didn't screw up any tags. Thanks, rnewton -- _ -- 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] 3621: documentation: features.conf.sample is unclear as to which parking options apply globally to all parking lots (1.8)
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3621/ --- (Updated June 23, 2014, 9:34 a.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 417076 Bugs: ASTERISK-23667 https://issues.asterisk.org/jira/browse/ASTERISK-23667 Repository: Asterisk Description --- The features.conf sample can be a bit confusing about what parking options can be set only in the general context, or both in the general context (for the default parking lot) and in other parking lot contexts. A bug was filed due to confusion and a little googling will show lots of other confused users. Despite some comments on the individual options, it still reads in a confusing way. In this patch I separate out those options with some headings in to attempt a better layout. I went ahead and modified other headings in the file, or added them to facilitate better visual scanning. Change to 11: https://reviewboard.asterisk.org/r/3622/ Diffs - /branches/1.8/configs/features.conf.sample 416556 Diff: https://reviewboard.asterisk.org/r/3621/diff/ Testing --- Sample file update, doesn't affect configuration. Only rearranged text, no addition or removal of options or contexts. So, no testing, other than looking at it! Thanks, rnewton -- _ -- 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] 3622: documentation: features.conf.sample is unclear as to which parking options apply globally to all parking lots (11)
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3622/ --- (Updated June 23, 2014, 9:35 a.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 417077 Bugs: ASTERISK-23667 https://issues.asterisk.org/jira/browse/ASTERISK-23667 Repository: Asterisk Description --- The features.conf sample can be a bit confusing about what parking options can be set only in the general context, or both in the general context (for the default parking lot) and in other parking lot contexts. A bug was filed due to confusion and a little googling will show lots of other confused users. Despite some comments on the individual options, it still reads in a confusing way. In this patch I separate out those options with some headings in to attempt a better layout. I went ahead and modified other headings in the file, or added them to facilitate better visual scanning. Change to 1.8: https://reviewboard.asterisk.org/r/3621/ Diffs - /branches/11/configs/features.conf.sample 416556 Diff: https://reviewboard.asterisk.org/r/3622/diff/ Testing --- Sample file update, doesn't affect configuration. Only rearranged text, no addition or removal of options or contexts. So, no testing, other than looking at it! Thanks, rnewton -- _ -- 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] 3621: documentation: features.conf.sample is unclear as to which parking options apply globally to all parking lots (1.8)
On June 18, 2014, 3:54 a.m., rmudgett wrote: /branches/1.8/configs/features.conf.sample, line 23 https://reviewboard.asterisk.org/r/3621/diff/2/?file=59773#file59773line23 You might want to have a standard default format for the options in this file for consistency. You likely should put the default note on its own line. We've already done some cleanup here going beyond the issue at hand, and doing this one looks to be pretty tedious considering most of the default notes have different formats throughout the file. I'm going to drop this one for now. My goal was just to make it more clear which options were global and which were per parking lot. On June 18, 2014, 3:54 a.m., rmudgett wrote: /branches/1.8/configs/features.conf.sample, line 99 https://reviewboard.asterisk.org/r/3621/diff/2/?file=59773#file59773line99 These lines could be removed since you have grouped them in the per parking lot section. This was quick, also removed the operates on all parking lots lines for the same reason. Plus added a note under the Parking Options heading. ; These options apply to all parking lots, including the default lot defined in ; the general context. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3621/#review12181 --- On June 17, 2014, 11:46 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3621/ --- (Updated June 17, 2014, 11:46 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-23667 https://issues.asterisk.org/jira/browse/ASTERISK-23667 Repository: Asterisk Description --- The features.conf sample can be a bit confusing about what parking options can be set only in the general context, or both in the general context (for the default parking lot) and in other parking lot contexts. A bug was filed due to confusion and a little googling will show lots of other confused users. Despite some comments on the individual options, it still reads in a confusing way. In this patch I separate out those options with some headings in to attempt a better layout. I went ahead and modified other headings in the file, or added them to facilitate better visual scanning. Change to 11: https://reviewboard.asterisk.org/r/3622/ Diffs - /branches/1.8/configs/features.conf.sample 416556 Diff: https://reviewboard.asterisk.org/r/3621/diff/ Testing --- Sample file update, doesn't affect configuration. Only rearranged text, no addition or removal of options or contexts. So, no testing, other than looking at it! Thanks, rnewton -- _ -- 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] 3621: documentation: features.conf.sample is unclear as to which parking options apply globally to all parking lots (1.8)
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3621/ --- (Updated June 19, 2014, 7:29 p.m.) Review request for Asterisk Developers. Changes --- Fixed richards comments Bugs: ASTERISK-23667 https://issues.asterisk.org/jira/browse/ASTERISK-23667 Repository: Asterisk Description --- The features.conf sample can be a bit confusing about what parking options can be set only in the general context, or both in the general context (for the default parking lot) and in other parking lot contexts. A bug was filed due to confusion and a little googling will show lots of other confused users. Despite some comments on the individual options, it still reads in a confusing way. In this patch I separate out those options with some headings in to attempt a better layout. I went ahead and modified other headings in the file, or added them to facilitate better visual scanning. Change to 11: https://reviewboard.asterisk.org/r/3622/ Diffs (updated) - /branches/1.8/configs/features.conf.sample 416556 Diff: https://reviewboard.asterisk.org/r/3621/diff/ Testing --- Sample file update, doesn't affect configuration. Only rearranged text, no addition or removal of options or contexts. So, no testing, other than looking at it! Thanks, rnewton -- _ -- 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] 3622: documentation: features.conf.sample is unclear as to which parking options apply globally to all parking lots (11)
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3622/ --- (Updated June 19, 2014, 7:30 p.m.) Review request for Asterisk Developers. Changes --- Fixed richards comments. Bugs: ASTERISK-23667 https://issues.asterisk.org/jira/browse/ASTERISK-23667 Repository: Asterisk Description --- The features.conf sample can be a bit confusing about what parking options can be set only in the general context, or both in the general context (for the default parking lot) and in other parking lot contexts. A bug was filed due to confusion and a little googling will show lots of other confused users. Despite some comments on the individual options, it still reads in a confusing way. In this patch I separate out those options with some headings in to attempt a better layout. I went ahead and modified other headings in the file, or added them to facilitate better visual scanning. Change to 1.8: https://reviewboard.asterisk.org/r/3621/ Diffs (updated) - /branches/11/configs/features.conf.sample 416556 Diff: https://reviewboard.asterisk.org/r/3622/diff/ Testing --- Sample file update, doesn't affect configuration. Only rearranged text, no addition or removal of options or contexts. So, no testing, other than looking at it! Thanks, rnewton -- _ -- 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] 3627: Update extensions.lua.sample with naming conflict guidance.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3627/#review12183 --- Ship it! Works for me. - rnewton On June 18, 2014, 1:29 a.m., George Joseph wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3627/ --- (Updated June 18, 2014, 1:29 a.m.) Review request for Asterisk Developers and rnewton. Bugs: asterisk-23844 https://issues.asterisk.org/jira/browse/asterisk-23844 Repository: Asterisk Description --- The sample extensions.lua was causing pbx_lua to fail to load when parsing 'app.goto(default, s, 1)' because in Lua 5.2, 'goto' is now a reserved word. Added the following guidance to extensions.lua.sample and changed 'app.goto(default, s, 1)' to 'app.['goto'](default, s, 1)'. -- Note about naming conflicts: -- Lua allows you to refer to table entries using the '.' notation, -- I.E. app.goto(something), only if the entry doesn't conflict with an Lua -- reserved word. In the 'goto' example, with Lua 5.1 or earlier, 'goto' is -- not a reserved word so you'd be calling the Asterisk dialplan application -- 'goto'. Lua 5.2 however, introduced the 'goto' control statement which -- makes 'goto' a reserved word. This casues the interpreter to fail parsing -- the file and pbx_lua.so will fail to load. The same applies to any use of -- Lua tables including extensions, channels and any tables you create. -- -- There are two ways around this: Since Lua is case-sensitive, you can use -- capitalized names, I.E. app.Goto(something) to refer to the Asterisk apps, -- functions, etc. Or you can use the full syntax, I.E. app[goto](something). -- Both syntaxes are backwards compatible with earlier Lua versions. To make -- your Lua dialplans easier to maintain and to reduce the chance of future -- conflicts you may want to use the app[goto](something) syntax for all -- table accesses. -- This patch will merge through to 11, 12 and trunk. The wiki needs to be update with the same info. Diffs - branches/1.8/configs/extensions.lua.sample 416556 Diff: https://reviewboard.asterisk.org/r/3627/diff/ Testing --- Made sure extensions.lua now loads correctly. 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] 3621: documentation: features.conf.sample is unclear as to which parking options apply globally to all parking lots (1.8)
On June 13, 2014, 10:46 p.m., rmudgett wrote: /branches/1.8/configs/features.conf.sample, lines 154-155 https://reviewboard.asterisk.org/r/3621/diff/1/?file=59658#file59658line154 The information in this note for parkext and parkpos should be put with the descriptions instead of here. I added those notes to the option descriptions *and* left the note in the original spot as well. It is a good reminder to a user who scrolls right to the example definition and doesn't read the individual option descriptions first. On June 13, 2014, 10:46 p.m., rmudgett wrote: /branches/1.8/configs/features.conf.sample, line 20 https://reviewboard.asterisk.org/r/3621/diff/1/?file=59658#file59658line20 You should group these global options to the various subgroups with a blank line separating the groups: pickup transfer transfer-sounds atxfer parking I used the groups, Pickup Options, Transfer Options and Parking Options. I didn't feel there was enough options to justify sub-groups beyond that. Thanks! - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3621/#review12153 --- On June 13, 2014, 8:44 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3621/ --- (Updated June 13, 2014, 8:44 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-23667 https://issues.asterisk.org/jira/browse/ASTERISK-23667 Repository: Asterisk Description --- The features.conf sample can be a bit confusing about what parking options can be set only in the general context, or both in the general context (for the default parking lot) and in other parking lot contexts. A bug was filed due to confusion and a little googling will show lots of other confused users. Despite some comments on the individual options, it still reads in a confusing way. In this patch I separate out those options with some headings in to attempt a better layout. I went ahead and modified other headings in the file, or added them to facilitate better visual scanning. Change to 11: https://reviewboard.asterisk.org/r/3622/ Diffs - /branches/1.8/configs/features.conf.sample 416208 Diff: https://reviewboard.asterisk.org/r/3621/diff/ Testing --- Sample file update, doesn't affect configuration. Only rearranged text, no addition or removal of options or contexts. So, no testing, other than looking at it! Thanks, rnewton -- _ -- 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] 3621: documentation: features.conf.sample is unclear as to which parking options apply globally to all parking lots (1.8)
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3621/ --- (Updated June 17, 2014, 11:46 p.m.) Review request for Asterisk Developers. Changes --- Fixed Richards comments and a fix trivial formatting issues or misspellings. Bugs: ASTERISK-23667 https://issues.asterisk.org/jira/browse/ASTERISK-23667 Repository: Asterisk Description --- The features.conf sample can be a bit confusing about what parking options can be set only in the general context, or both in the general context (for the default parking lot) and in other parking lot contexts. A bug was filed due to confusion and a little googling will show lots of other confused users. Despite some comments on the individual options, it still reads in a confusing way. In this patch I separate out those options with some headings in to attempt a better layout. I went ahead and modified other headings in the file, or added them to facilitate better visual scanning. Change to 11: https://reviewboard.asterisk.org/r/3622/ Diffs (updated) - /branches/1.8/configs/features.conf.sample 416556 Diff: https://reviewboard.asterisk.org/r/3621/diff/ Testing --- Sample file update, doesn't affect configuration. Only rearranged text, no addition or removal of options or contexts. So, no testing, other than looking at it! Thanks, rnewton -- _ -- 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] 3622: documentation: features.conf.sample is unclear as to which parking options apply globally to all parking lots (11)
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3622/ --- (Updated June 17, 2014, 11:46 p.m.) Review request for Asterisk Developers. Changes --- Fixed Richards comments(from https://reviewboard.asterisk.org/r/3621/) and a fix trivial formatting issues or misspellings. Bugs: ASTERISK-23667 https://issues.asterisk.org/jira/browse/ASTERISK-23667 Repository: Asterisk Description --- The features.conf sample can be a bit confusing about what parking options can be set only in the general context, or both in the general context (for the default parking lot) and in other parking lot contexts. A bug was filed due to confusion and a little googling will show lots of other confused users. Despite some comments on the individual options, it still reads in a confusing way. In this patch I separate out those options with some headings in to attempt a better layout. I went ahead and modified other headings in the file, or added them to facilitate better visual scanning. Change to 1.8: https://reviewboard.asterisk.org/r/3621/ Diffs (updated) - /branches/11/configs/features.conf.sample 416556 Diff: https://reviewboard.asterisk.org/r/3622/diff/ Testing --- Sample file update, doesn't affect configuration. Only rearranged text, no addition or removal of options or contexts. So, no testing, other than looking at it! Thanks, rnewton -- _ -- 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] 3621: documentation: features.conf.sample is unclear as to which parking options apply globally to all parking lots
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3621/ --- Review request for Asterisk Developers. Bugs: ASTERISK-23667 https://issues.asterisk.org/jira/browse/ASTERISK-23667 Repository: Asterisk Description --- The features.conf sample can be a bit confusing about what parking options can be set only in the general context, or both in the general context (for the default parking lot) and in other parking lot contexts. A bug was filed due to confusion and a little googling will show lots of other confused users. Despite some comments on the individual options, it still reads in a confusing way. In this patch I separate out those options with some headings in to attempt a better layout. I went ahead and modified other headings in the file, or added them to facilitate better visual scanning. Will link the review for 11 here, since it is a little different. Diffs - /branches/1.8/configs/features.conf.sample 416208 Diff: https://reviewboard.asterisk.org/r/3621/diff/ Testing --- Sample file update, doesn't affect configuration. Only rearranged text, no addition or removal of options or contexts. So, no testing, other than looking at it! Thanks, rnewton -- _ -- 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] 3621: documentation: features.conf.sample is unclear as to which parking options apply globally to all parking lots (1.8)
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3621/ --- (Updated June 13, 2014, 8:43 p.m.) Review request for Asterisk Developers. Summary (updated) - documentation: features.conf.sample is unclear as to which parking options apply globally to all parking lots (1.8) Bugs: ASTERISK-23667 https://issues.asterisk.org/jira/browse/ASTERISK-23667 Repository: Asterisk Description --- The features.conf sample can be a bit confusing about what parking options can be set only in the general context, or both in the general context (for the default parking lot) and in other parking lot contexts. A bug was filed due to confusion and a little googling will show lots of other confused users. Despite some comments on the individual options, it still reads in a confusing way. In this patch I separate out those options with some headings in to attempt a better layout. I went ahead and modified other headings in the file, or added them to facilitate better visual scanning. Will link the review for 11 here, since it is a little different. Diffs - /branches/1.8/configs/features.conf.sample 416208 Diff: https://reviewboard.asterisk.org/r/3621/diff/ Testing --- Sample file update, doesn't affect configuration. Only rearranged text, no addition or removal of options or contexts. So, no testing, other than looking at it! Thanks, rnewton -- _ -- 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] 3621: documentation: features.conf.sample is unclear as to which parking options apply globally to all parking lots (1.8)
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3621/ --- (Updated June 13, 2014, 8:44 p.m.) Review request for Asterisk Developers. Changes --- edited desc Bugs: ASTERISK-23667 https://issues.asterisk.org/jira/browse/ASTERISK-23667 Repository: Asterisk Description (updated) --- The features.conf sample can be a bit confusing about what parking options can be set only in the general context, or both in the general context (for the default parking lot) and in other parking lot contexts. A bug was filed due to confusion and a little googling will show lots of other confused users. Despite some comments on the individual options, it still reads in a confusing way. In this patch I separate out those options with some headings in to attempt a better layout. I went ahead and modified other headings in the file, or added them to facilitate better visual scanning. Change to 11: https://reviewboard.asterisk.org/r/3622/ Diffs - /branches/1.8/configs/features.conf.sample 416208 Diff: https://reviewboard.asterisk.org/r/3621/diff/ Testing --- Sample file update, doesn't affect configuration. Only rearranged text, no addition or removal of options or contexts. So, no testing, other than looking at it! Thanks, rnewton -- _ -- 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] 3622: documentation: features.conf.sample is unclear as to which parking options apply globally to all parking lots (11)
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3622/ --- Review request for Asterisk Developers. Bugs: ASTERISK-23667 https://issues.asterisk.org/jira/browse/ASTERISK-23667 Repository: Asterisk Description --- The features.conf sample can be a bit confusing about what parking options can be set only in the general context, or both in the general context (for the default parking lot) and in other parking lot contexts. A bug was filed due to confusion and a little googling will show lots of other confused users. Despite some comments on the individual options, it still reads in a confusing way. In this patch I separate out those options with some headings in to attempt a better layout. I went ahead and modified other headings in the file, or added them to facilitate better visual scanning. Change to 1.8: https://reviewboard.asterisk.org/r/3621/ Diffs - /branches/11/configs/features.conf.sample 415997 Diff: https://reviewboard.asterisk.org/r/3622/diff/ Testing --- Sample file update, doesn't affect configuration. Only rearranged text, no addition or removal of options or contexts. So, no testing, other than looking at it! Thanks, rnewton -- _ -- 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] 3610: main/pbx - documentation - enhance 'core show hints' and 'core show hint' help text
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3610/ --- (Updated June 12, 2014, 4:15 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 415998 Bugs: ASTERISK-23764 https://issues.asterisk.org/jira/browse/ASTERISK-23764 Repository: Asterisk Description --- ASTERISK-23764 is already closed, but item 3 in the list didn't get done on another issue, so I'm doing it now. Previous output was: centosclean*CLI core show help core show hint Usage: core show hint exten List registered hint centosclean*CLI core show help core show hints Usage: core show hints List registered hints The new output for the branches (main diff): centosclean*CLI core show help core show hint Usage: core show hint exten List registered hint. Hint details are shown in four columns. In order from left to right, they are: 1. Hint extension URI. 2. Mapped device state identifiers. 3. Current extension state. The aggregate of mapped device states. 4. Watchers - number of subscriptions and other entities watching this hint. centosclean*CLI core show help core show hints Usage: core show hints List registered hints. Hint details are shown in four columns. In order from left to right, they are: 1. Hint extension URI. 2. Mapped device state identifiers. 3. Current extension state. The aggregate of mapped device states. 4. Watchers - number of subscriptions and other entities watching this hint. The new output for trunk (attached diff) takes into account https://reviewboard.asterisk.org/r/3604/ and https://reviewboard.asterisk.org/r/3611/ newtonr-laptop*CLI core show help core show hint Usage: core show hint exten List registered hint. Hint details are shown in five columns. In order from left to right, they are: 1. Hint extension URI. 2. List of mapped device or presence state identifiers. 3. Current extension state. The aggregate of mapped device states. 4. Current presence state for the mapped presence state provider. 5. Watchers - number of subscriptions and other entities watching this hint. newtonr-laptop*CLI newtonr-laptop*CLI core show help core show hints Usage: core show hints List registered hints. Hint details are shown in five columns. In order from left to right, they are: 1. Hint extension URI. 2. List of mapped device or presence state identifiers. 3. Current extension state. The aggregate of mapped device states. 4. Current presence state for the mapped presence state provider. 5. Watchers - number of subscriptions and other entities watching this hint. The diff covers the help for 'core show hint' and 'core show hints' in 1.8,11,12. The attached patch covers the same for Trunk. Diffs - /branches/1.8/main/pbx.c 415710 Diff: https://reviewboard.asterisk.org/r/3610/diff/ Testing --- Tested output in 1.8 and Trunk. File Attachments patch for trunk https://reviewboard.asterisk.org/media/uploaded/files/2014/06/11/0eb22a12-e297-4ec3-8191-fd1774a50f33__coreshowhints_trunk.patch Thanks, rnewton -- _ -- 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] 3610: main/pbx - documentation - enhance 'core show hints' and 'core show hint' help text
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3610/ --- Review request for Asterisk Developers. Bugs: ASTERISK-23764 https://issues.asterisk.org/jira/browse/ASTERISK-23764 Repository: Asterisk Description --- ASTERISK-23764 is already closed, but item 3 in the list didn't get done on another issue, so I'm doing it now. Previous output was: centosclean*CLI core show help core show hint Usage: core show hint exten List registered hint centosclean*CLI core show help core show hints Usage: core show hints List registered hints The new output for the branches (main diff): centosclean*CLI core show help core show hint Usage: core show hint exten List registered hint. Hint details are shown in four columns. In order from left to right, they are: 1. Hint extension URI. 2. Mapped device state identifiers. 3. Current extension state. The aggregate of mapped device states. 4. Watchers - number of subscriptions and other entities watching this hint. centosclean*CLI core show help core show hints Usage: core show hints List registered hints. Hint details are shown in four columns. In order from left to right, they are: 1. Hint extension URI. 2. Mapped device state identifiers. 3. Current extension state. The aggregate of mapped device states. 4. Watchers - number of subscriptions and other entities watching this hint. The new output for trunk (attached diff) takes into account https://reviewboard.asterisk.org/r/3604/ and https://reviewboard.asterisk.org/r/3611/ newtonr-laptop*CLI core show help core show hint Usage: core show hint exten List registered hint. Hint details are shown in five columns. In order from left to right, they are: 1. Hint extension URI. 2. List of mapped device or presence state identifiers. 3. Current extension state. The aggregate of mapped device states. 4. Current presence state for the mapped presence state provider. 5. Watchers - number of subscriptions and other entities watching this hint. newtonr-laptop*CLI newtonr-laptop*CLI core show help core show hints Usage: core show hints List registered hints. Hint details are shown in five columns. In order from left to right, they are: 1. Hint extension URI. 2. List of mapped device or presence state identifiers. 3. Current extension state. The aggregate of mapped device states. 4. Current presence state for the mapped presence state provider. 5. Watchers - number of subscriptions and other entities watching this hint. The diff covers the help for 'core show hint' and 'core show hints' in 1.8,11,12. The attached patch covers the same for Trunk. Diffs - /branches/1.8/main/pbx.c 415710 Diff: https://reviewboard.asterisk.org/r/3610/diff/ Testing --- Tested output in 1.8 and Trunk. File Attachments patch for trunk https://reviewboard.asterisk.org/media/uploaded/files/2014/06/11/0eb22a12-e297-4ec3-8191-fd1774a50f33__coreshowhints_trunk.patch Thanks, rnewton -- _ -- 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] 3611: CLI: correct presence state display
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3611/#review12129 --- Ship it! Looks like it works to me! The presence state now looks accurate and reflects what I see in presencestate list newtonr-laptop*CLI core show hints -= Registered Asterisk Dial Plan Hints =- 6002@from-internal : PJSIP/6002,CustomPre State:Idle Presence:available Watchers 1 --- Name: 'CustomPresence:Alice' --- State: 'available' --- Subtype: '' --- Message: '' --- Base64 Encoded: 'No' Plus 'core show hint' has the new column. newtonr-laptop*CLI core show hint 6002 6002@from-internal : PJSIP/6002,CustomPre State:Idle Presence:available Watchers 1 1 hint matching extension 6002 - rnewton On June 11, 2014, 6 p.m., Scott Griepentrog wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3611/ --- (Updated June 11, 2014, 6 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-23858 https://issues.asterisk.org/jira/browse/ASTERISK-23858 Repository: Asterisk Description --- This change adds the presence column to the core show hint X output, and changes the presence string conversion to use the correct function. Diffs - /trunk/main/pbx.c 415698 Diff: https://reviewboard.asterisk.org/r/3611/diff/ Testing --- Thanks, Scott Griepentrog -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3572: cli_aliases: Add 'pjsip reload' alias because it's nice. Plus, another for 11 syntax compatability.
On May 28, 2014, 11:57 p.m., Paul Belanger wrote: I'm all for adding the pjsip reload, but the context changes in the template seem unnecessary and potentially break systems that depend on the default config files. Matt Jordan wrote: Do you have an alternative to propose for naming Asterisk 12+ CLI aliases? Those syntax backwards compatibility templates are commented out in the config sample file. Only the friendly template is enabled. No system should be dependent on those. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3572/#review11989 --- On May 28, 2014, 8:14 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3572/ --- (Updated May 28, 2014, 8:14 p.m.) Review request for Asterisk Developers. Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-23654 https://issues.asterisk.org/jira/browse/https://issues.asterisk.org/jira/browse/ASTERISK-23654 Repository: Asterisk Description --- 1. changed naming of included alias templates to avoid confusion between version names. For example, asterisk12 was for asterisk 1.2, so I changed it to asterisk_1dot2, so that later we can use asterisk_12 for Asterisk 12. 2. added alias for features reload to the template for Asterisk 11 style syntax template, as features reload was removed in 12, but you can still do module reload features 3. added alias for pjsip reload to the friendly template. It is shorter than module reload res_pjsip.so and if some are like me; I constantly forget that reloading chan_pjsip doesn't parse config. Remembering pjsip reload is just easier. Diffs - /branches/12/configs/cli_aliases.conf.sample 414779 Diff: https://reviewboard.asterisk.org/r/3572/diff/ Testing --- Tested the added aliases. Thanks, rnewton -- _ -- 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] 3572: cli_aliases: Add 'pjsip reload' alias because it's nice. Plus, others for 11 syntax compatability.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3572/ --- Review request for Asterisk Developers. Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-23654 https://issues.asterisk.org/jira/browse/https://issues.asterisk.org/jira/browse/ASTERISK-23654 Repository: Asterisk Description --- 1. changed naming of included alias templates to avoid confusion between version names. For example, asterisk12 was for asterisk 1.2, so I changed it to asterisk_1dot2, so that later we can use asterisk_12 for Asterisk 12. 2. added alias for features reload to the template for Asterisk 11 style syntax template, as features reload was removed in 12, but you can still do module reload features 3. added alias for pjsip reload to the friendly template. It is shorter than module reload res_pjsip.so and if some are like me; I constantly forget that reloading chan_pjsip doesn't parse config. Remembering pjsip reload is just easier. Diffs - /branches/12/configs/cli_aliases.conf.sample 414779 Diff: https://reviewboard.asterisk.org/r/3572/diff/ Testing --- Tested the added aliases. Thanks, rnewton -- _ -- 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] 3572: cli_aliases: Add 'pjsip reload' alias because it's nice. Plus, another for 11 syntax compatability.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3572/ --- (Updated May 28, 2014, 8:14 p.m.) Review request for Asterisk Developers. Summary (updated) - cli_aliases: Add 'pjsip reload' alias because it's nice. Plus, another for 11 syntax compatability. Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-23654 https://issues.asterisk.org/jira/browse/https://issues.asterisk.org/jira/browse/ASTERISK-23654 Repository: Asterisk Description --- 1. changed naming of included alias templates to avoid confusion between version names. For example, asterisk12 was for asterisk 1.2, so I changed it to asterisk_1dot2, so that later we can use asterisk_12 for Asterisk 12. 2. added alias for features reload to the template for Asterisk 11 style syntax template, as features reload was removed in 12, but you can still do module reload features 3. added alias for pjsip reload to the friendly template. It is shorter than module reload res_pjsip.so and if some are like me; I constantly forget that reloading chan_pjsip doesn't parse config. Remembering pjsip reload is just easier. Diffs - /branches/12/configs/cli_aliases.conf.sample 414779 Diff: https://reviewboard.asterisk.org/r/3572/diff/ Testing --- Tested the added aliases. Thanks, rnewton -- _ -- 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] 3542: Documentation: Wiki page for Maintenance and Upgrades, including sub pages.
On May 15, 2014, 9:22 p.m., Scott Griepentrog wrote: In updating or upgrading asterisk, you use the terms first, then later explain what they mean. It would be better to define the terms first, then use them. Good point. Fixed this on https://wiki.asterisk.org/wiki/display/AST/Updating+or+Upgrading+Asterisk - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3542/#review11896 --- On May 15, 2014, 9:13 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3542/ --- (Updated May 15, 2014, 9:13 p.m.) Review request for Asterisk Developers. Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-23396 None Description --- We needed a wiki page that at least gives an overview of tasks associated with maintaining an Asterisk system. Such as backups, updating or upgrading and other miscellaneous maintenance related tasks. I wrote a few pages to get this started. Please give it a quick read and a ship it, if it looks good. I'm sure I said something dumb or inaccurate somewhere. :) https://wiki.asterisk.org/wiki/display/AST/Maintenance+and+Upgrades Plus two sub-pages: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Backups https://wiki.asterisk.org/wiki/display/AST/Updating+or+Upgrading+Asterisk I'd like to know about: * Things to expand on * Things I said too much about or went out of scope on * Things missing If you already have wiki edit access feel free to edit typos/logic errors straight-away, otherwise just report them on here. Diffs - Diff: https://reviewboard.asterisk.org/r/3542/diff/ Testing --- Thanks, rnewton -- _ -- 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] 3542: Documentation: Wiki page for Maintenance and Upgrades, including sub pages.
On May 16, 2014, 5:24 p.m., Scott Griepentrog wrote: Nitpick: There is an extra space in the Asterisk support life-cycle. Fixed thanks - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3542/#review11906 --- On May 15, 2014, 9:13 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3542/ --- (Updated May 15, 2014, 9:13 p.m.) Review request for Asterisk Developers. Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-23396 None Description --- We needed a wiki page that at least gives an overview of tasks associated with maintaining an Asterisk system. Such as backups, updating or upgrading and other miscellaneous maintenance related tasks. I wrote a few pages to get this started. Please give it a quick read and a ship it, if it looks good. I'm sure I said something dumb or inaccurate somewhere. :) https://wiki.asterisk.org/wiki/display/AST/Maintenance+and+Upgrades Plus two sub-pages: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Backups https://wiki.asterisk.org/wiki/display/AST/Updating+or+Upgrading+Asterisk I'd like to know about: * Things to expand on * Things I said too much about or went out of scope on * Things missing If you already have wiki edit access feel free to edit typos/logic errors straight-away, otherwise just report them on here. Diffs - Diff: https://reviewboard.asterisk.org/r/3542/diff/ Testing --- Thanks, rnewton -- _ -- 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] 3542: Documentation: Wiki page for Maintenance and Upgrades, including sub pages.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3542/ --- (Updated May 16, 2014, 5:30 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-23396 None Description --- We needed a wiki page that at least gives an overview of tasks associated with maintaining an Asterisk system. Such as backups, updating or upgrading and other miscellaneous maintenance related tasks. I wrote a few pages to get this started. Please give it a quick read and a ship it, if it looks good. I'm sure I said something dumb or inaccurate somewhere. :) https://wiki.asterisk.org/wiki/display/AST/Maintenance+and+Upgrades Plus two sub-pages: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Backups https://wiki.asterisk.org/wiki/display/AST/Updating+or+Upgrading+Asterisk I'd like to know about: * Things to expand on * Things I said too much about or went out of scope on * Things missing If you already have wiki edit access feel free to edit typos/logic errors straight-away, otherwise just report them on here. Diffs - Diff: https://reviewboard.asterisk.org/r/3542/diff/ Testing --- Thanks, rnewton -- _ -- 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] 3542: Documentation: Wiki page for Maintenance and Upgrades, including sub pages.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3542/ --- (Updated May 15, 2014, 9:13 p.m.) Review request for Asterisk Developers. Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-23396 None Description --- We needed a wiki page that at least gives an overview of tasks associated with maintaining an Asterisk system. Such as backups, updating or upgrading and other miscellaneous maintenance related tasks. I wrote a few pages to get this started. Please give it a quick read and a ship it, if it looks good. I'm sure I said something dumb or inaccurate somewhere. :) https://wiki.asterisk.org/wiki/display/AST/Maintenance+and+Upgrades Plus two sub-pages: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Backups https://wiki.asterisk.org/wiki/display/AST/Updating+or+Upgrading+Asterisk I'd like to know about: * Things to expand on * Things I said too much about or went out of scope on * Things missing If you already have wiki edit access feel free to edit typos/logic errors straight-away, otherwise just report them on here. Diffs - Diff: https://reviewboard.asterisk.org/r/3542/diff/ Testing --- Thanks, rnewton -- _ -- 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] 3542: Documentation: Wiki page for Maintenance and Upgrades, including sub pages.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3542/ --- Review request for Asterisk Developers. Bugs: ASTERISK-23396 None Description --- We needed a wiki page that at least gives an overview of tasks associated with maintaining an Asterisk system. Such as backups, updating or upgrading and other miscellaneous maintenance related tasks. I wrote a few pages to get this started. Please give it a quick read and a ship it, if it looks good. I'm sure I said something dumb or inaccurate somewhere. :) https://wiki.asterisk.org/wiki/display/AST/Maintenance+and+Upgrades Plus two sub-pages: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Backups https://wiki.asterisk.org/wiki/display/AST/Updating+or+Upgrading+Asterisk I'd like to know about: * Things to expand on * Things I said too much about or went out of scope on * Things missing If you already have wiki edit access feel free to edit typos/logic errors straight-away, otherwise just report them on here. Diffs - Diff: https://reviewboard.asterisk.org/r/3542/diff/ Testing --- Thanks, rnewton -- _ -- 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] 3464: Sounds: Various new sound sets were missing from the makefile and menuselect options
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3464/ --- (Updated April 18, 2014, 12:12 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 412585 Bugs: ASTERISK-23550 https://issues.asterisk.org/jira/browse/ASTERISK-23550 Repository: Asterisk Description --- Apparently back when we added the IT,en_GB,en_AU sound sets, we didn't get the menuselect stuff modified for it all. (I'm pretty sure I just didn't realize what I had to modify) Along the way I fixed some other minor problems. The main diff does the following: In sounds/Makefile 1 Adds and moves some lines necessary for the en_GB core set. I'm just following how the other sets are defined here. 2 removes the ES extra sounds related lines as we don't have ES extra sound sets. In sounds/sounds.xml 3 Adds support_level definitions to all the sound sets as we have these defined in 11,12,Trunk, but not in 1.8 4 Adds member definitons for EN_AU, EN_GB, IT for core sound sets, and EN_GB in extra sound sets The attached 11plus.patch is the same patch, but without item #3 above, as that is fine in 11,12,Trunk. Diffs - /branches/1.8/sounds/sounds.xml 412484 /branches/1.8/sounds/Makefile 412484 Diff: https://reviewboard.asterisk.org/r/3464/diff/ Testing --- In 1.8 and 12, configured in dev mode, built, installed, verified sound packages selected in menuselect get pulled down and extracted to their various directories. Tested with a random selection of formats from each language set of en_AU,en_GB, IT. File Attachments Patch for 11,12,Trunk https://reviewboard.asterisk.org/media/uploaded/files/2014/04/17/4421ac84-bf9f-417d-9449-a582ee90430d__asterisk23550_11plus.patch Thanks, rnewton -- _ -- 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] 3188: format_wav: enhancing log message Not a wav file to be clear on what is supported
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3188/
---
(Updated Feb. 6, 2014, 1:29 p.m.)
Status
--
This change has been marked as submitted.
Review request for Asterisk Developers.
Bugs: ASTERISK-22310
https://issues.asterisk.org/jira/browse/ASTERISK-22310
Repository: Asterisk
Description
---
From Jim's report:
The following error message lies. Many things are WAV files which do not
contain 1 in the format field.
if (ltohs(format) != 1) {
ast_log(LOG_WARNING, Not a wav file %d\n, ltohs(format));
Given how unusual the .WAV vs. .wav distinction is I'd go for: Not a supported
wav file format (%d). Only PCM encoded versions are supported with a lower-case
'.wav' extension.
Which would be helpful.
The message I've suggested in the patch is:
Not a supported wav file format (%d). Only PCM encoded, 16 bit, mono, 8kHz
files are supported with a lowercase '.wav' extension.\n
Please let me know:
1) Is the message content accurate according to format_wav.c ?
2) Is the message more understandable and less ambiguous than the original?
Diffs
-
/branches/12/formats/format_wav.c 407338
Diff: https://reviewboard.asterisk.org/r/3188/diff/
Testing
---
Thanks,
rnewton
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Re: [asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3180/ --- (Updated Feb. 6, 2014, 8 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Repository: Asterisk Description --- There is some nuance to naming configuration sections. In the long term I'll hope that configuration section names become a bit more arbitrary; in the short term help me make sure this documentation patch clarifies things. Diffs - /branches/12/configs/pjsip.conf.sample 407338 Diff: https://reviewboard.asterisk.org/r/3180/diff/ Testing --- Only adding informational text to the pjsip.conf.sample file. Thanks, rnewton -- _ -- 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] 2990: Documentation: Clarify x Option In chan_spy
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/2990/#review10765 --- Ship it! Ship It! - rnewton On Nov. 2, 2013, 1:30 p.m., Michael Young wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/2990/ --- (Updated Nov. 2, 2013, 1:30 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-22661 https://issues.asterisk.org/jira/browse/ASTERISK-22661 Repository: Asterisk Description --- When using the x option (specify a DTMF digit to exit the application), it is not obvious in the documentation that this only works when spying on a channel. If a channel being used to spy on other channels is waiting to connect to a channel or is no longer attached to a channel, the DTMF is ignored. As noted on the issue tracker, since there are workarounds available and this is a rarely used option we are opting for a documentation change here. This patch helps to clarify the documentation. Diffs - /branches/1.8/apps/app_chanspy.c 402263 Diff: https://reviewboard.asterisk.org/r/2990/diff/ Testing --- Thanks, Michael Young -- _ -- 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] 3136: cli: pjsip show endpoint endpoint shows allow/disallow codecs the same
On Feb. 5, 2014, 7:37 p.m., rnewton wrote: Ship It! Tested command, output looks good and makes sense to me. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3136/#review10766 --- On Feb. 3, 2014, 8:18 p.m., Scott Griepentrog wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3136/ --- (Updated Feb. 3, 2014, 8:18 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-23092 https://issues.asterisk.org/jira/browse/ASTERISK-23092 Repository: Asterisk Description --- WAS: Insert a ! prefix in the display of endpoint disallow value. Result is: disallow : !(ulaw|alaw) NOW: Remove the disallow option from generated lists, while still accepting it from a configuration file. Diffs - /branches/12/res/res_pjsip/pjsip_configuration.c 407196 /branches/12/main/sorcery.c 407196 /branches/12/main/config_options.c 407196 /branches/12/include/asterisk/config_options.h 407196 Diff: https://reviewboard.asterisk.org/r/3136/diff/ Testing --- Ran command and checked output. Thanks, Scott Griepentrog -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint endpoint shows allow/disallow codecs the same
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3136/#review10766 --- Ship it! Ship It! - rnewton On Feb. 3, 2014, 8:18 p.m., Scott Griepentrog wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3136/ --- (Updated Feb. 3, 2014, 8:18 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-23092 https://issues.asterisk.org/jira/browse/ASTERISK-23092 Repository: Asterisk Description --- WAS: Insert a ! prefix in the display of endpoint disallow value. Result is: disallow : !(ulaw|alaw) NOW: Remove the disallow option from generated lists, while still accepting it from a configuration file. Diffs - /branches/12/res/res_pjsip/pjsip_configuration.c 407196 /branches/12/main/sorcery.c 407196 /branches/12/main/config_options.c 407196 /branches/12/include/asterisk/config_options.h 407196 Diff: https://reviewboard.asterisk.org/r/3136/diff/ Testing --- Ran command and checked output. Thanks, Scott Griepentrog -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3180/ --- (Updated Feb. 5, 2014, 9:10 p.m.) Review request for Asterisk Developers. Changes --- Another attempt. Should resolve the mentioned issues. Repository: Asterisk Description --- There is some nuance to naming configuration sections. In the long term I'll hope that configuration section names become a bit more arbitrary; in the short term help me make sure this documentation patch clarifies things. Diffs (updated) - /branches/12/configs/pjsip.conf.sample 407338 Diff: https://reviewboard.asterisk.org/r/3180/diff/ Testing --- Only adding informational text to the pjsip.conf.sample file. Thanks, rnewton -- _ -- 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] 3188: format_wav: enhancing log message Not a wav file to be clear on what is supported
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3188/
---
Review request for Asterisk Developers.
Summary (updated)
-
format_wav: enhancing log message Not a wav file to be clear on what is
supported
Bugs: ASTERISK-22310
https://issues.asterisk.org/jira/browse/ASTERISK-22310
Repository: Asterisk
Description (updated)
---
From Jim's report:
The following error message lies. Many things are WAV files which do not
contain 1 in the format field.
if (ltohs(format) != 1) {
ast_log(LOG_WARNING, Not a wav file %d\n, ltohs(format));
Given how unusual the .WAV vs. .wav distinction is I'd go for: Not a supported
wav file format (%d). Only PCM encoded versions are supported with a lower-case
'.wav' extension.
Which would be helpful.
The message I've suggested in the patch is:
Not a supported wav file format (%d). Only PCM encoded, 16 bit, mono, 8kHz
files are supported with a lowercase '.wav' extension.\n
Please let me know:
1) Is the message content accurate according to format_wav.c ?
2) Is the message more understandable and less ambiguous than the original?
Diffs
-
Diff: https://reviewboard.asterisk.org/r/3188/diff/
Testing
---
Thanks,
rnewton
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Re: [asterisk-dev] [Code Review] 3188: format_wav: enhancing log message Not a wav file to be clear on what is supported
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3188/
---
(Updated Feb. 5, 2014, 10:14 p.m.)
Review request for Asterisk Developers.
Changes
---
Diff wasn't uploaded for whatever reason.
Bugs: ASTERISK-22310
https://issues.asterisk.org/jira/browse/ASTERISK-22310
Repository: Asterisk
Description
---
From Jim's report:
The following error message lies. Many things are WAV files which do not
contain 1 in the format field.
if (ltohs(format) != 1) {
ast_log(LOG_WARNING, Not a wav file %d\n, ltohs(format));
Given how unusual the .WAV vs. .wav distinction is I'd go for: Not a supported
wav file format (%d). Only PCM encoded versions are supported with a lower-case
'.wav' extension.
Which would be helpful.
The message I've suggested in the patch is:
Not a supported wav file format (%d). Only PCM encoded, 16 bit, mono, 8kHz
files are supported with a lowercase '.wav' extension.\n
Please let me know:
1) Is the message content accurate according to format_wav.c ?
2) Is the message more understandable and less ambiguous than the original?
Diffs (updated)
-
/branches/12/formats/format_wav.c 407338
Diff: https://reviewboard.asterisk.org/r/3188/diff/
Testing
---
Thanks,
rnewton
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[asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3180/ --- Review request for Asterisk Developers. Repository: Asterisk Description --- There is some nuance to naming configuration sections. In the long term I'll hope that configuration section names become a bit more arbitrary; in the short term help me make sure this documentation patch clarifies things. Diffs - /branches/12/configs/pjsip.conf.sample 407338 Diff: https://reviewboard.asterisk.org/r/3180/diff/ Testing --- Only adding informational text to the pjsip.conf.sample file. Thanks, rnewton -- _ -- 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] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3180/ --- (Updated Feb. 4, 2014, 11:12 p.m.) Review request for Asterisk Developers. Changes --- Ended up rewriting and adding a bit more text to fix the issues. Repository: Asterisk Description --- There is some nuance to naming configuration sections. In the long term I'll hope that configuration section names become a bit more arbitrary; in the short term help me make sure this documentation patch clarifies things. Diffs (updated) - /branches/12/configs/pjsip.conf.sample 407338 Diff: https://reviewboard.asterisk.org/r/3180/diff/ Testing --- Only adding informational text to the pjsip.conf.sample file. Thanks, rnewton -- _ -- 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] 3086: pjsip.conf.sample update: improve documentation of pjsip endpoints behind NAT and update for snake case change
On Dec. 20, 2013, 1:56 a.m., Matt Jordan wrote: branches/12/configs/pjsip.conf.sample, line 435 https://reviewboard.asterisk.org/r/3086/diff/1/?file=49880#file49880line435 May as well align the comment with the others Everything in that section is generated by my script, I adjusted this line manually, but I'll poke the script to see if we can get the comment alignment better later on. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3086/#review10469 --- On Dec. 20, 2013, 12:18 a.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3086/ --- (Updated Dec. 20, 2013, 12:18 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-23004 https://issues.asterisk.org/jira/browse/ASTERISK-23004 Repository: Asterisk Description --- Added another NAT example to pjsip.conf.sample. We had a few mentions of NAT configuration throughout the sample, but I added another for a little bit more clarity. Additionally many pjsip options were affected by the change to snake case, so I fixed any instances of those options in pjsip.conf. I regenerated the config option list (at the bottom of the file) from a new xml config doc dump, so all the snake case changes should be reflected there, as well as any other changes to those options. Diffs - branches/12/configs/pjsip.conf.sample 404396 Diff: https://reviewboard.asterisk.org/r/3086/diff/ Testing --- Only documentation outside of source code, doesn't appear to break anything. Thanks, rnewton -- _ -- 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] 3086: pjsip.conf.sample update: improve documentation of pjsip endpoints behind NAT and update for snake case change
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3086/ --- (Updated Dec. 20, 2013, 3:28 p.m.) Review request for Asterisk Developers. Changes --- Updated diff after fixing Matt's findings. Bugs: ASTERISK-23004 https://issues.asterisk.org/jira/browse/ASTERISK-23004 Repository: Asterisk Description --- Added another NAT example to pjsip.conf.sample. We had a few mentions of NAT configuration throughout the sample, but I added another for a little bit more clarity. Additionally many pjsip options were affected by the change to snake case, so I fixed any instances of those options in pjsip.conf. I regenerated the config option list (at the bottom of the file) from a new xml config doc dump, so all the snake case changes should be reflected there, as well as any other changes to those options. Diffs (updated) - branches/12/configs/pjsip.conf.sample 404403 Diff: https://reviewboard.asterisk.org/r/3086/diff/ Testing --- Only documentation outside of source code, doesn't appear to break anything. Thanks, rnewton -- _ -- 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] 3086: pjsip.conf.sample update: improve documentation of pjsip endpoints behind NAT and update for snake case change
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3086/ --- (Updated Dec. 20, 2013, 7:19 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Bugs: ASTERISK-23004 https://issues.asterisk.org/jira/browse/ASTERISK-23004 Repository: Asterisk Description --- Added another NAT example to pjsip.conf.sample. We had a few mentions of NAT configuration throughout the sample, but I added another for a little bit more clarity. Additionally many pjsip options were affected by the change to snake case, so I fixed any instances of those options in pjsip.conf. I regenerated the config option list (at the bottom of the file) from a new xml config doc dump, so all the snake case changes should be reflected there, as well as any other changes to those options. Diffs - branches/12/configs/pjsip.conf.sample 404403 Diff: https://reviewboard.asterisk.org/r/3086/diff/ Testing --- Only documentation outside of source code, doesn't appear to break anything. Thanks, rnewton -- _ -- 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] 3043: pjsip: CLI commands
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3043/#review10476 --- For pjsip show channels The output looks like: *CLI pjsip show channels Channel: ChannelId State. Time(sec) Codec: Codec Exten: DialedExten. CLCID: ConnectedLineCID... = Channel: Up 43 Codec: (ulaw) Exten: 6002CLCID: Channel: Up 43 Codec: (ulaw) Exten: CLCID: RustyONE 6001 It appears that the ChannelID is not populating. *CLI core show channels Channel Location State Application(Data) PJSIP/6002-0001 (None) Up AppDial((Outgoing Line)) PJSIP/6001- (None) Up Dial(PJSIP/6002,15) 2 active channels 1 active call 1 call processed - rnewton On Dec. 20, 2013, 4:24 a.m., George Joseph wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3043/ --- (Updated Dec. 20, 2013, 4:24 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-22610 https://issues.asterisk.org/jira/browse/ASTERISK-22610 Repository: Asterisk Description --- Implements the following cli commands: pjsip list aors pjsip list auths pjsip list channels pjsip list contacts pjsip list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show channels pjsip show endpoint(s) Also... Minor modifications made to the AMI command implementations to facilitate reuse. New function ast_variable_list_sort added to config.c and config.h to implement variable list sorting. New api ast_sip_for_each_identify added to module res_pjsip_endpoint_identifier_ip. Required new files res_pjsip_endpoint_identifier_ip.h and res_pjsip_endpoint_identifier_ip.exports.in. Implementation of the summary statistics is still pending. I'm sure there will be lots of feedback here. :) Diffs - branches/12/res/res_pjsip_registrar.c 404396 branches/12/res/res_pjsip_endpoint_identifier_ip.exports.in PRE-CREATION branches/12/res/res_pjsip_endpoint_identifier_ip.c 404396 branches/12/res/res_pjsip/pjsip_configuration.c 404396 branches/12/res/res_pjsip/pjsip_cli.c PRE-CREATION branches/12/res/res_pjsip/location.c 404396 branches/12/res/res_pjsip/include/res_pjsip_private.h 404396 branches/12/res/res_pjsip/config_auth.c 404396 branches/12/main/sorcery.c 404396 branches/12/main/config.c 404396 branches/12/include/asterisk/sorcery.h 404396 branches/12/include/asterisk/res_pjsip_endpoint_identifier_ip.h PRE-CREATION branches/12/include/asterisk/res_pjsip_cli.h PRE-CREATION branches/12/include/asterisk/res_pjsip.h 404396 branches/12/include/asterisk/config.h 404396 Diff: https://reviewboard.asterisk.org/r/3043/diff/ Testing --- I've gone through all the combinations of actions and objects and made sure at least the correct objects are returned, there are no segfaults, errors, etc. I *think* the formatting is correct but I'll need some feedback. 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] 3043: pjsip: CLI commands
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3043/#review10479 --- For pjsip show endpoints, a formatting issue: Endpoint/CID State. Channels. I/OAuth: AuthId/UserName... Aor: Aor MaxContact Contact: Aor/ContactUri... Status RTT(ms).. Identify: MatchList. Channel: ChannelId.. State. Time(sec) Codec: Codec Exten: DialedExten... CLCID: ConnectedLineCID... = 6001In use 1 of inf InAuth: 6001/6001 Aor: 6001 2 Contact: 6001/sip:6001@10.24.18.16:5060;obAvail 3.739 Channel: PJSIP/6001-/Dial Up497 Codec: (ulaw) Exten: 6002 CLCID: For the Channel: line, the State and Time(sec) values are out of their columns and the State is pushed up against the Channel ID to it's left. - rnewton On Dec. 20, 2013, 4:24 a.m., George Joseph wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3043/ --- (Updated Dec. 20, 2013, 4:24 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-22610 https://issues.asterisk.org/jira/browse/ASTERISK-22610 Repository: Asterisk Description --- Implements the following cli commands: pjsip list aors pjsip list auths pjsip list channels pjsip list contacts pjsip list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show channels pjsip show endpoint(s) Also... Minor modifications made to the AMI command implementations to facilitate reuse. New function ast_variable_list_sort added to config.c and config.h to implement variable list sorting. New api ast_sip_for_each_identify added to module res_pjsip_endpoint_identifier_ip. Required new files res_pjsip_endpoint_identifier_ip.h and res_pjsip_endpoint_identifier_ip.exports.in. Implementation of the summary statistics is still pending. I'm sure there will be lots of feedback here. :) Diffs - branches/12/res/res_pjsip_registrar.c 404396 branches/12/res/res_pjsip_endpoint_identifier_ip.exports.in PRE-CREATION branches/12/res/res_pjsip_endpoint_identifier_ip.c 404396 branches/12/res/res_pjsip/pjsip_configuration.c 404396 branches/12/res/res_pjsip/pjsip_cli.c PRE-CREATION branches/12/res/res_pjsip/location.c 404396 branches/12/res/res_pjsip/include/res_pjsip_private.h 404396 branches/12/res/res_pjsip/config_auth.c 404396 branches/12/main/sorcery.c 404396 branches/12/main/config.c 404396 branches/12/include/asterisk/sorcery.h 404396 branches/12/include/asterisk/res_pjsip_endpoint_identifier_ip.h PRE-CREATION branches/12/include/asterisk/res_pjsip_cli.h PRE-CREATION branches/12/include/asterisk/res_pjsip.h 404396 branches/12/include/asterisk/config.h 404396 Diff: https://reviewboard.asterisk.org/r/3043/diff/ Testing --- I've gone through all the combinations of actions and objects and made sure at least the correct objects are returned, there are no segfaults, errors, etc. I *think* the formatting is correct but I'll need some feedback. 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] 3043: pjsip: CLI commands
On Dec. 20, 2013, 7:35 p.m., rnewton wrote: For pjsip show endpoints, a formatting issue: Endpoint/CID State. Channels. I/OAuth: AuthId/UserName... Aor: Aor MaxContact Contact: Aor/ContactUri... Status RTT(ms).. Identify: MatchList. Channel: ChannelId.. State. Time(sec) Codec: Codec Exten: DialedExten... CLCID: ConnectedLineCID... = 6001In use 1 of inf InAuth: 6001/6001 Aor: 6001 2 Contact: 6001/sip:6001@10.24.18.16:5060;obAvail 3.739 Channel: PJSIP/6001-/Dial Up497 Codec: (ulaw) Exten: 6002 CLCID: For the Channel: line, the State and Time(sec) values are out of their columns and the State is pushed up against the Channel ID to it's left. This alignment issue also appears to apply to pjsip show channels - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3043/#review10479 --- On Dec. 20, 2013, 4:24 a.m., George Joseph wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3043/ --- (Updated Dec. 20, 2013, 4:24 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-22610 https://issues.asterisk.org/jira/browse/ASTERISK-22610 Repository: Asterisk Description --- Implements the following cli commands: pjsip list aors pjsip list auths pjsip list channels pjsip list contacts pjsip list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show channels pjsip show endpoint(s) Also... Minor modifications made to the AMI command implementations to facilitate reuse. New function ast_variable_list_sort added to config.c and config.h to implement variable list sorting. New api ast_sip_for_each_identify added to module res_pjsip_endpoint_identifier_ip. Required new files res_pjsip_endpoint_identifier_ip.h and res_pjsip_endpoint_identifier_ip.exports.in. Implementation of the summary statistics is still pending. I'm sure there will be lots of feedback here. :) Diffs - branches/12/res/res_pjsip_registrar.c 404396 branches/12/res/res_pjsip_endpoint_identifier_ip.exports.in PRE-CREATION branches/12/res/res_pjsip_endpoint_identifier_ip.c 404396 branches/12/res/res_pjsip/pjsip_configuration.c 404396 branches/12/res/res_pjsip/pjsip_cli.c PRE-CREATION branches/12/res/res_pjsip/location.c 404396 branches/12/res/res_pjsip/include/res_pjsip_private.h 404396 branches/12/res/res_pjsip/config_auth.c 404396 branches/12/main/sorcery.c 404396 branches/12/main/config.c 404396 branches/12/include/asterisk/sorcery.h 404396 branches/12/include/asterisk/res_pjsip_endpoint_identifier_ip.h PRE-CREATION branches/12/include/asterisk/res_pjsip_cli.h PRE-CREATION branches/12/include/asterisk/res_pjsip.h 404396 branches/12/include/asterisk/config.h 404396 Diff: https://reviewboard.asterisk.org/r/3043/diff/ Testing --- I've gone through all the combinations of actions and objects and made sure at least the correct objects are returned, there are no segfaults, errors, etc. I *think* the formatting is correct but I'll need some feedback. 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] 3043: pjsip: CLI commands
On Dec. 20, 2013, 7:35 p.m., rnewton wrote: For pjsip show endpoints, a formatting issue: Endpoint/CID State. Channels. I/OAuth: AuthId/UserName... Aor: Aor MaxContact Contact: Aor/ContactUri... Status RTT(ms).. Identify: MatchList. Channel: ChannelId.. State. Time(sec) Codec: Codec Exten: DialedExten... CLCID: ConnectedLineCID... = 6001In use 1 of inf InAuth: 6001/6001 Aor: 6001 2 Contact: 6001/sip:6001@10.24.18.16:5060;obAvail 3.739 Channel: PJSIP/6001-/Dial Up497 Codec: (ulaw) Exten: 6002 CLCID: For the Channel: line, the State and Time(sec) values are out of their columns and the State is pushed up against the Channel ID to it's left. rnewton wrote: This alignment issue also appears to apply to pjsip show channels and pjsip list channels - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3043/#review10479 --- On Dec. 20, 2013, 4:24 a.m., George Joseph wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3043/ --- (Updated Dec. 20, 2013, 4:24 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-22610 https://issues.asterisk.org/jira/browse/ASTERISK-22610 Repository: Asterisk Description --- Implements the following cli commands: pjsip list aors pjsip list auths pjsip list channels pjsip list contacts pjsip list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show channels pjsip show endpoint(s) Also... Minor modifications made to the AMI command implementations to facilitate reuse. New function ast_variable_list_sort added to config.c and config.h to implement variable list sorting. New api ast_sip_for_each_identify added to module res_pjsip_endpoint_identifier_ip. Required new files res_pjsip_endpoint_identifier_ip.h and res_pjsip_endpoint_identifier_ip.exports.in. Implementation of the summary statistics is still pending. I'm sure there will be lots of feedback here. :) Diffs - branches/12/res/res_pjsip_registrar.c 404396 branches/12/res/res_pjsip_endpoint_identifier_ip.exports.in PRE-CREATION branches/12/res/res_pjsip_endpoint_identifier_ip.c 404396 branches/12/res/res_pjsip/pjsip_configuration.c 404396 branches/12/res/res_pjsip/pjsip_cli.c PRE-CREATION branches/12/res/res_pjsip/location.c 404396 branches/12/res/res_pjsip/include/res_pjsip_private.h 404396 branches/12/res/res_pjsip/config_auth.c 404396 branches/12/main/sorcery.c 404396 branches/12/main/config.c 404396 branches/12/include/asterisk/sorcery.h 404396 branches/12/include/asterisk/res_pjsip_endpoint_identifier_ip.h PRE-CREATION branches/12/include/asterisk/res_pjsip_cli.h PRE-CREATION branches/12/include/asterisk/res_pjsip.h 404396 branches/12/include/asterisk/config.h 404396 Diff: https://reviewboard.asterisk.org/r/3043/diff/ Testing --- I've gone through all the combinations of actions and objects and made sure at least the correct objects are returned, there are no segfaults, errors, etc. I *think* the formatting is correct but I'll need some feedback. 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] 3043: pjsip: CLI commands
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3043/#review10482 --- The state reflected in the command outputs remains Invalid and does not update until the endpoint has made at least one call. Then it will change to Not in use *CLI pjsip show endpoints snip 6001Not in use 0 of inf InAuth: 6001/6001 Aor: 6001 2 Contact: 6001/sip:6001@10.24.18.16:5060;obUnknown nan 6002Invalid 0 of inf InAuth: 6002/6002 Aor: 6002 2 Contact: 6002/sip:6002@10.24.18.138:5060;ob Unknown nan In this example, both 6001 and 6002 are registered, but 6001 has made a call and then hung up. 6002 has not made a call yet. - rnewton On Dec. 20, 2013, 4:24 a.m., George Joseph wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3043/ --- (Updated Dec. 20, 2013, 4:24 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-22610 https://issues.asterisk.org/jira/browse/ASTERISK-22610 Repository: Asterisk Description --- Implements the following cli commands: pjsip list aors pjsip list auths pjsip list channels pjsip list contacts pjsip list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show channels pjsip show endpoint(s) Also... Minor modifications made to the AMI command implementations to facilitate reuse. New function ast_variable_list_sort added to config.c and config.h to implement variable list sorting. New api ast_sip_for_each_identify added to module res_pjsip_endpoint_identifier_ip. Required new files res_pjsip_endpoint_identifier_ip.h and res_pjsip_endpoint_identifier_ip.exports.in. Implementation of the summary statistics is still pending. I'm sure there will be lots of feedback here. :) Diffs - branches/12/res/res_pjsip_registrar.c 404396 branches/12/res/res_pjsip_endpoint_identifier_ip.exports.in PRE-CREATION branches/12/res/res_pjsip_endpoint_identifier_ip.c 404396 branches/12/res/res_pjsip/pjsip_configuration.c 404396 branches/12/res/res_pjsip/pjsip_cli.c PRE-CREATION branches/12/res/res_pjsip/location.c 404396 branches/12/res/res_pjsip/include/res_pjsip_private.h 404396 branches/12/res/res_pjsip/config_auth.c 404396 branches/12/main/sorcery.c 404396 branches/12/main/config.c 404396 branches/12/include/asterisk/sorcery.h 404396 branches/12/include/asterisk/res_pjsip_endpoint_identifier_ip.h PRE-CREATION branches/12/include/asterisk/res_pjsip_cli.h PRE-CREATION branches/12/include/asterisk/res_pjsip.h 404396 branches/12/include/asterisk/config.h 404396 Diff: https://reviewboard.asterisk.org/r/3043/diff/ Testing --- I've gone through all the combinations of actions and objects and made sure at least the correct objects are returned, there are no segfaults, errors, etc. I *think* the formatting is correct but I'll need some feedback. 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] 3043: pjsip: CLI commands
On Dec. 20, 2013, 7:35 p.m., rnewton wrote: For pjsip show endpoints, a formatting issue: Endpoint/CID State. Channels. I/OAuth: AuthId/UserName... Aor: Aor MaxContact Contact: Aor/ContactUri... Status RTT(ms).. Identify: MatchList. Channel: ChannelId.. State. Time(sec) Codec: Codec Exten: DialedExten... CLCID: ConnectedLineCID... = 6001In use 1 of inf InAuth: 6001/6001 Aor: 6001 2 Contact: 6001/sip:6001@10.24.18.16:5060;obAvail 3.739 Channel: PJSIP/6001-/Dial Up497 Codec: (ulaw) Exten: 6002 CLCID: For the Channel: line, the State and Time(sec) values are out of their columns and the State is pushed up against the Channel ID to it's left. rnewton wrote: This alignment issue also appears to apply to pjsip show channels rnewton wrote: and pjsip list channels I updated to latest for the pjsip-cli branch and these all appear fixed - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3043/#review10479 --- On Dec. 20, 2013, 4:24 a.m., George Joseph wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3043/ --- (Updated Dec. 20, 2013, 4:24 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-22610 https://issues.asterisk.org/jira/browse/ASTERISK-22610 Repository: Asterisk Description --- Implements the following cli commands: pjsip list aors pjsip list auths pjsip list channels pjsip list contacts pjsip list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show channels pjsip show endpoint(s) Also... Minor modifications made to the AMI command implementations to facilitate reuse. New function ast_variable_list_sort added to config.c and config.h to implement variable list sorting. New api ast_sip_for_each_identify added to module res_pjsip_endpoint_identifier_ip. Required new files res_pjsip_endpoint_identifier_ip.h and res_pjsip_endpoint_identifier_ip.exports.in. Implementation of the summary statistics is still pending. I'm sure there will be lots of feedback here. :) Diffs - branches/12/res/res_pjsip_registrar.c 404396 branches/12/res/res_pjsip_endpoint_identifier_ip.exports.in PRE-CREATION branches/12/res/res_pjsip_endpoint_identifier_ip.c 404396 branches/12/res/res_pjsip/pjsip_configuration.c 404396 branches/12/res/res_pjsip/pjsip_cli.c PRE-CREATION branches/12/res/res_pjsip/location.c 404396 branches/12/res/res_pjsip/include/res_pjsip_private.h 404396 branches/12/res/res_pjsip/config_auth.c 404396 branches/12/main/sorcery.c 404396 branches/12/main/config.c 404396 branches/12/include/asterisk/sorcery.h 404396 branches/12/include/asterisk/res_pjsip_endpoint_identifier_ip.h PRE-CREATION branches/12/include/asterisk/res_pjsip_cli.h PRE-CREATION branches/12/include/asterisk/res_pjsip.h 404396 branches/12/include/asterisk/config.h 404396 Diff: https://reviewboard.asterisk.org/r/3043/diff/ Testing --- I've gone through all the combinations of actions and objects and made sure at least the correct objects are returned, there are no segfaults, errors, etc. I *think* the formatting is correct but I'll need some feedback. 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
[asterisk-dev] [Code Review] 3086: pjsip.conf.sample update: improve documentation of pjsip endpoints behind NAT and update for snake case change
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3086/ --- Review request for Asterisk Developers. Bugs: ASTERISK-23004 https://issues.asterisk.org/jira/browse/ASTERISK-23004 Repository: Asterisk Description --- Added another NAT example to pjsip.conf.sample. We had a few mentions of NAT configuration throughout the sample, but I added another for a little bit more clarity. Additionally many pjsip options were affected by the change to snake case, so I fixed any instances of those options in pjsip.conf. I regenerated the config option list (at the bottom of the file) from a new xml config doc dump, so all the snake case changes should be reflected there, as well as any other changes to those options. Diffs - branches/12/configs/pjsip.conf.sample 404396 Diff: https://reviewboard.asterisk.org/r/3086/diff/ Testing --- Only documentation outside of source code, doesn't appear to break anything. Thanks, rnewton -- _ -- 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
