Re: [asterisk-dev] Asterisk 16 Parking Lot Full behavior.

he > public. > > On Thu, Dec 20, 2018 at 10:57 AM Jonathan Rose motorolasolutions.com > <https://urldefense.proofpoint.com/v2/url?u=http-3A__lists.digium.com_mailman_listinfo_asterisk-2Ddev=DwMCaQ=q3cDpHe1hF8lXU5EFjNM_A=50uagQBTpQAKCx3KjAwJcMd6ygCPToAyDAxH5

Re: [asterisk-dev] Asterisk 16 Parking Lot Full behavior.

t; > n,ParkAndAnnounce(${DYNAMICPARKINGNAME},c(callParking_timeout,s,1)t(${HASH(lot_info,park_time)}),silence/1:PARKED,Local/${DIALEDPEERNUMBER}@parkCall_do_park) > same => n,NoOp("After blind parkAndAnnounce") > > I can provide any more information anyone requests. > > On Wed,

Re: [asterisk-dev] Asterisk 16 Parking Lot Full behavior.

That's a bit of a flawed approach. The highest parking space can be occupied while other spots are open. Parked calls don't get shuffled to lower spots as lower numbered parking spots are freed up. Plus there are multiple modes for selecting the parking space for a call. That would be a safer

Re: [asterisk-dev] Asterisk 16 Parking Lot Full behavior.

>ParkAndAnnounce I actually missed this detail from the original email. Yeah, ParkAndAnnounce might also behave a lot differently since it's going to involve the origination of channels. Honestly instead of relying on that you could simply have some script listen to manager for ParkedCall events

Re: [asterisk-dev] Asterisk 16 Parking Lot Full behavior.

This might work better in general if you use DTMF feature transfers to park instead of trying to use an on-phone transfer button to send the call to a park extension. Asterisk has internal logic to check an extension that you send a parked call to and will rely on res_parking functionality to

Re: [asterisk-dev] Channels in a bridge

Using ast_channel_get_bridge(struct ast_channel *chan), you can get a reference for the bridge. The channel needs to be locked before you use that function and you'll need to cleanup the bridge reference when you are done with it. From there you can use the bridge reference to get the list of

Re: [asterisk-dev] Anyone interested in having file/line number shown in dialplan show output?

On Wed, Dec 7, 2016 at 1:42 PM, Corey Farrell wrote: > Overall I like it. I think leaving pbx_ael out of this is best as it > might not be straight forward. Anyone who wants pbx_ael to provide > information about config sources can create a follow-up patch once the > core

[asterisk-dev] Anyone interested in having file/line number shown in dialplan show output?

I've been working through a lot of complicated dialplan setups lately, mostly written in AEL. Doing so has left me painfully aware of the fact that it can sometimes be hard to trace a dialplan through its various extensions. More so in pbx_ael of course where what you write isn't necessarily

Re: [asterisk-dev] Measuring Asterisk performance

On Wed, Sep 28, 2016 at 4:30 AM, Nitesh Bansal wrote: > Hello, > > I'm trying to understand if I could use a system metric like load average, > cpu usage... to > decide if Asterisk is overloaded and if it is overloaded, I would like to > stop routing the traffic > to

Re: [asterisk-dev] Advice for having a bug fixed

On Thu, Jul 21, 2016 at 10:16 AM, Jonathan Rose < jonathan.r...@motorolasolutions.com> wrote: > > On Thu, Jul 21, 2016 at 2:30 AM, Leandro Dardini <ldard...@gmail.com> > wrote: > >> Hello, >> I am looking for an advice for having a bug fixed. I am talking

Re: [asterisk-dev] Advice for having a bug fixed

On Thu, Jul 21, 2016 at 2:30 AM, Leandro Dardini wrote: > Hello, > I am looking for an advice for having a bug fixed. I am talking about > ASTERISK-25468 > > https://issues.asterisk.org/jira/browse/ASTERISK-25468 >

Re: [asterisk-dev] Development of asterisk 1.4.23 Can we please get some development?

On Thu, Jul 14, 2016 at 3:27 PM, Loren Tedford wrote: > First off i want to say thanks for the fast replies and want to say no I > don't plan on putting out a job offer.. But was just interested in some > help in general of putting in at least the basics we require into >

Re: [asterisk-dev] Development of asterisk 1.4.23 Can we please get some development?

> > > Is it possible that anyone in the development team would help us make our > version of asterisk better.. Or so I should say the original version of > asterisk better. > > Not a snowball's chance in hell. > So.. Is anyone in the active development group interested in helping us get past

Re: [asterisk-dev] [BOUNTY] Bug Fixes

> > > Well, there are a couple of reasons. The first is that I don't really know > what it's going to take to fix these issues, I just have a decent idea of > what > kind of story point estimates we would have given these issues back when I > was doing this kind of work and I also have a decent

Re: [asterisk-dev] [BOUNTY] Bug Fixes

up three other things. > > Please keep this in mind when you're so flippant about what the effort is > going to be. > > Regards, > Phil Mickelson > > > On Mon, Jul 11, 2016 at 11:21 AM, Jonathan Rose < > jonathan.r...@motorolasolutions.com> wrote: > >&g

Re: [asterisk-dev] [BOUNTY] Bug Fixes

On Mon, Jul 11, 2016 at 6:37 AM, Ross Beer wrote: > Hi Sean, > > > $500 in total as they shouldn't need too much work. > > > Regards, > > > Ross > > > -- > *From:* asterisk-dev-boun...@lists.digium.com < > asterisk-dev-boun...@lists.digium.com>

Re: [asterisk-dev] Asterisk Load Performance

On Tue, Jul 5, 2016 at 3:43 PM, Michael Petruzzello < michael.petruzze...@civi.com> wrote: > On Wed, Jun 29 at 11:14:04 AM, Richard Mudgett

Re: [asterisk-dev] [Code Review] 4577: res_pjsip_t38: T38 fax fails when using authentication with PJSIP sender

this locally with my two Asterisk machines in the above scenario. I'll be linking the test after putting it on Gerrit. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev

Re: [asterisk-dev] [Code Review] 4577: res_pjsip_t38: T38 fax fails when using authentication with PJSIP sender

in the above scenario. I'll be linking the test after putting it on Gerrit. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update

[asterisk-dev] Change in testsuite[master]: Testsuite: New test for FAX via PJSIP T38 with authentication

Jonathan Rose has uploaded a new patch set (#4). Change subject: Testsuite: New test for FAX via PJSIP T38 with authentication .. Testsuite: New test for FAX via PJSIP T38 with authentication Add a test for PJSIP t38

Re: [asterisk-dev] [Code Review] 4577: res_pjsip_t38: T38 fax fails when using authentication with PJSIP sender

/#review15069 --- On April 3, 2015, 12:03 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4577

Re: [asterisk-dev] [Code Review] 4577: res_pjsip_t38: T38 fax fails when using authentication with PJSIP sender

. - Jonathan --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4577/#review15075 --- On April 6, 2015, 11:14 a.m., Jonathan Rose wrote

Re: [asterisk-dev] [Code Review] 4577: res_pjsip_t38: T38 fax fails when using authentication with PJSIP sender

and passes with the patch. I also tested this locally with my two Asterisk machines in the above scenario. I'll be linking the test after putting it on Gerrit. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http

[asterisk-dev] Change in testsuite[master]: Add a test for PJSIP t38 with authentication based on normal...

: testsuite Gerrit-Branch: master Gerrit-Owner: Jonathan Rose jr...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com -- _ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-dev] Change in testsuite[master]: Address review feedback

Jonathan Rose has uploaded a new change for review. https://gerrit.asterisk.org/28 Change subject: Address review feedback .. Address review feedback Change-Id: If37cf20857ae3c0b35e0637a0a2cb7e7d6226df6 --- M tests/fax/pjsip

[asterisk-dev] Change in testsuite[master]: Add a test for PJSIP t38 with authentication based on normal...

Jonathan Rose has uploaded a new patch set (#2). Change subject: Add a test for PJSIP t38 with authentication based on normal t38 test The test will start two instances of Asterisk. The first will originate a PJSIP call with authentication to the second using an extension that will run sendFax

Re: [asterisk-dev] [Code Review] 4577: res_pjsip_t38: T38 fax fails when using authentication with PJSIP sender

://reviewboard.asterisk.org/r/4577/#comment25722 s/for the channel/for the endpoint/ - Jonathan Rose On April 3, 2015, 12:03 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https

[asterisk-dev] Change in testsuite[master]: Add a test for PJSIP t38 with authentication based on normal...

Jonathan Rose has abandoned this change. Change subject: Add a test for PJSIP t38 with authentication based on normal t38 test .. Abandoned I'm abandoning this one since the patch got mangled. Go over to c/28 -- To view

Re: [asterisk-dev] [Code Review] 4577: res_pjsip_t38: T38 fax fails when using authentication with PJSIP sender

scenario. I'll be linking the test after putting it on Gerrit. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit

[asterisk-dev] Change in testsuite[master]: Add a test for PJSIP t38 with authentication based on normal...

, visit https://gerrit.asterisk.org/22 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: newpatchset Gerrit-Change-Id: Id8fd9683dc1b61e7b1afd2b6ede857921beebb88 Gerrit-PatchSet: 2 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Jonathan Rose jr...@digium.com

[asterisk-dev] Change in testsuite[master]: Add a test for PJSIP t38 with authentication based on normal...

://gerrit.asterisk.org/22 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: newpatchset Gerrit-Change-Id: Id8fd9683dc1b61e7b1afd2b6ede857921beebb88 Gerrit-PatchSet: 3 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Jonathan Rose jr...@digium.com Gerrit-Reviewer: Ashley

[asterisk-dev] Change in testsuite[master]: Add a test for PJSIP t38 with authentication based on normal...

Jonathan Rose has posted comments on this change. Change subject: Add a test for PJSIP t38 with authentication based on normal t38 test .. Patch Set 1: (2 comments) https://gerrit.asterisk.org/#/c/22/1/tests/fax/pjsip

Re: [asterisk-dev] [Code Review] 4577: res_pjsip_t38: T38 fax fails when using authentication with PJSIP sender

it again. rmudgett wrote: It is specific framehooks that cannot be attached twice. It is not a general requirement/property of the framehook concept. Kevin Harwell wrote: Just curious, but what's a case where a framehook should be attached twice? Jonathan Rose wrote: You

Re: [asterisk-dev] [Code Review] 4577: res_pjsip_t38: T38 fax fails when using authentication with PJSIP sender

--- On April 2, 2015, 2:08 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4577

[asterisk-dev] [Code Review] 4577: res_pjsip_t38: T38 fax fails when using authentication with PJSIP sender

be linking the test after putting it on Gerrit. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-dev] [Code Review] 4488: Super Awesome Company: Phase 1 - Patch 2 - Outside Connectivity!

stuff and I suppose doing a quick glance to see if you missed anything else in testing. - Jonathan Rose On March 24, 2015, 4:53 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https

Re: [asterisk-dev] [Code Review] 4504: SAC: Add conferences for employees / employees+customers

/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

Re: [asterisk-dev] [Code Review] 4503: SAC: Configure customer advocate/sales queues

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

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/#review14850 --- - Jonathan Rose On March 24, 2015, 4:53 p.m., rnewton wrote

Re: [asterisk-dev] [Code Review] 4488: Super Awesome Company: Phase 1 - Patch 2 - Outside Connectivity!

://reviewboard.asterisk.org/r/4488/#comment25456 Was this intended to be commented out like this? - Jonathan Rose On March 24, 2015, 4:53 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https

Re: [asterisk-dev] [Code Review] 4510: app_confbridge (13): file playback blocks dtmf

://reviewboard.asterisk.org/r/4510/#comment25397 No parameters list. - Jonathan Rose On March 19, 2015, 4:59 p.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4510

Re: [asterisk-dev] [Code Review] 4511: Audit ast_pjsip_rdata_get_endpoint() usage for ref leaks.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4511/#review14749 --- Ship it! Ship It! - Jonathan Rose On March 17, 2015, 10:04

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/#review14730 --- On March 16, 2015, 10:37 a.m., Jonathan Rose wrote

[asterisk-dev] [Code Review] 4503: SAC: Configure customer advocate/sales queues

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

[asterisk-dev] [Code Review] 4504: SAC: Add conferences for employees / employees+customers

--- 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

Re: [asterisk-dev] [Code Review] 4488: Super Awesome Company: Phase 1 - Patch 2 - Outside Connectivity!

://reviewboard.asterisk.org/r/4488/#comment25241 Since we already know what the numbers for these queues are going to be, you could probably go ahead and put the GOTO operations in there and stub them out. - Jonathan Rose On March 13, 2015, 9:32 a.m., rnewton wrote

Re: [asterisk-dev] [Code Review] 4373: Manager Action ModuleLoad gives incorrect response when used to reload modules

: No such module. CLI output showed the module was actually reloaded successfully. After Patch: Action: ModuleLoad LoadType: reload Module: pbx_config.so Response: Success Message: Module reloaded. Thanks, Jonathan Rose

[asterisk-dev] [Code Review] 4373: Manager Action ModuleLoad gives incorrect response when used to reload modules

: pbx_config.so Response: Success Message: Module reloaded. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-dev] [Code Review] 4373: Manager Action ModuleLoad gives incorrect response when used to reload modules

: pbx_config.so Response: Error Message: No such module. CLI output showed the module was actually reloaded successfully. After Patch: Action: ModuleLoad LoadType: reload Module: pbx_config.so Response: Success Message: Module reloaded. Thanks, Jonathan Rose

Re: [asterisk-dev] [Code Review] 4050: Add ability for Channel Drivers to provide Presence State information

your presence provider function to retrieve the presence state of the channel 4 - Change the presence state of the channel 5 - Check presence state again 6 - release the channel - Jonathan Rose On Oct. 16, 2014, 11:26 p.m., gareth wrote

Re: [asterisk-dev] [Code Review] 4120: res_pjsip_acl: contact ACL permits are being interpreted incorrectly

have to reload pjsip after startup in order for the ACL to work... that seems like a bug surely. In any event, with the patch the device successfully registers. Without the patch, the registration is blocked by the ACL. Thanks, Jonathan Rose

Re: [asterisk-dev] [Code Review] 4109: Documentation: CDR unanswered behavior

the definition... at least in those scenarios. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-dev] [Code Review] 4120: res_pjsip_acl: contact ACL permits are being interpreted incorrectly

successfully registers. Without the patch, the registration is blocked by the ACL. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE

Re: [asterisk-dev] [Code Review] 4117: Fix building chan_phone on big endian systems

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4117/#review13603 --- Ship it! Ship It! - Jonathan Rose On Oct. 28, 2014, 12:46

Re: [asterisk-dev] [Code Review] 4085: ExtensionStatus: Add additional documentation describing the ExtensionStatus event

value Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

[asterisk-dev] [Code Review] 4109: Documentation: CDR unanswered behavior

--- I checked some simple calling scenarios involve app_dial in one case and call files in another to make sure that the behavior didn't differ from the definition... at least in those scenarios. Thanks, Jonathan Rose

Re: [asterisk-dev] [Code Review] 4085: ExtensionStatus: Add additional documentation describing the ExtensionStatus event

., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4085/ --- (Updated Oct. 15, 2014, 2:01 p.m.) Review

Re: [asterisk-dev] [Code Review] 4085: ExtensionStatus: Add additional documentation describing the ExtensionStatus event

devices are RINGING and one or more devices are INUSE Hold - All devices are ONHOLD. InUseHold - All devices are ONHOLD and one or more devices are INUSE Unknown - None of the above descriptions matched the status value Thanks, Jonathan Rose

[asterisk-dev] [Code Review] 4085: ExtensionStatus: Add additional documentation describing the ExtensionStatus event

devices are ONHOLD. InUseHold - All devices are ONHOLD and one or more devices are INUSE Unknown - None of the above descriptions matched the status value Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-dev] [Code Review] 4075: parking/tests: Running res_parking unit tests would cause assertions and possibly a crash due to attempting to play MOH on a channel with no formats

occurred with the patch in place. 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

Re: [asterisk-dev] [Code Review] 4071: scheduler: Fix a bug introduced by adding a delete flag to scheduled tasks

and without having chan_sip break down, I retested without the patch and quickly ran into the problem again. I think it's safe to say that I got it. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-dev] [Code Review] 4075: parking/tests: Running res_parking unit tests would cause assertions and possibly a crash due to attempting to play MOH on a channel with no formats

(updated) - /branches/13/res/parking/parking_tests.c 425404 Diff: https://reviewboard.asterisk.org/r/4075/diff/ Testing --- Ran tests prior to patch and got assertions. Assertions no longer occurred with the patch in place. Thanks, Jonathan Rose

Re: [asterisk-dev] [Code Review] 4075: parking/tests: Running res_parking unit tests would cause assertions and possibly a crash due to attempting to play MOH on a channel with no formats

no longer occurred with the patch in place. 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

[asterisk-dev] [Code Review] 4075: parking/tests: Running res_parking unit tests would cause assertions and possibly a crash due to attempting to play MOH on a channel with no formats

/parking_tests.c 425404 Diff: https://reviewboard.asterisk.org/r/4075/diff/ Testing --- Ran tests prior to patch and got assertions. Assertions no longer occurred with the patch in place. Thanks, Jonathan Rose

Re: [asterisk-dev] [Code Review] 4075: parking/tests: Running res_parking unit tests would cause assertions and possibly a crash due to attempting to play MOH on a channel with no formats

of these things. - Jonathan --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4075/#review13506 --- On Oct. 13, 2014, 3:59 p.m., Jonathan Rose

Re: [asterisk-dev] [Code Review] 4075: parking/tests: Running res_parking unit tests would cause assertions and possibly a crash due to attempting to play MOH on a channel with no formats

, 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] 4075: parking/tests: Running res_parking unit tests would cause assertions and possibly a crash due to attempting to play MOH on a channel with no formats

, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

[asterisk-dev] [Code Review] 4071: scheduler: Fix a bug introduced by adding a delete flag to scheduled tasks

got it. 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] 4071: scheduler: Fix a bug introduced by adding a delete flag to scheduled tasks

into the problem again. I think it's safe to say that I got it. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-dev] [Code Review] 4020: RLS Tests: off nominal tests for lists of lists (MWI and presence)

for things like eventlist support to guarantee that the tests that expected eventlist support failed if they didn't have it and that the tests that expected not to have eventlist support and had it also failed. Thanks, Jonathan Rose

Re: [asterisk-dev] [Code Review] 4013: Alembic: Add 'outgoing' enum value to sippeers directmedia enumerator

/diff/ Testing --- upgraded, checked to see if I could put 'outgoing' in the directmedia column. Downgraded, upgraded again to make sure things didn't explode. All on a postgres database Thanks, Jonathan Rose

Re: [asterisk-dev] [Code Review] 4046: audiohooks: Reevaluate the bridge technology when an audiohook is added or removed.

the behavior I noted on the issue. Ship it, ship it good. - Jonathan Rose On Oct. 2, 2014, 4:35 p.m., rmudgett wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4046

Re: [asterisk-dev] [Code Review] 4045: chan_sip: process_sdp leaks on an error path

. - Jonathan Rose On Oct. 1, 2014, 9:43 p.m., Corey Farrell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4045

Re: [asterisk-dev] [Code Review] 4017: chan_pjsip: Don't attempt to apply formats if there aren't any capabilities defined when creating a new channel

-, AGI Exit Status: ) in new stack Oooh, now isn't that clean. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update

Re: [asterisk-dev] [Code Review] 4020: RLS Tests: off nominal tests for lists of lists (MWI and presence)

/4020/#review13407 --- On Sept. 24, 2014, 5:28 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4020

Re: [asterisk-dev] [Code Review] 4020: RLS Tests: off nominal tests for lists of lists (MWI and presence)

that the tests that expected eventlist support failed if they didn't have it and that the tests that expected not to have eventlist support and had it also failed. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation

Re: [asterisk-dev] [Code Review] 4017: chan_pjsip: Don't attempt to apply formats if there aren't any capabilities defined when creating a new channel

] NoOp(PJSIP/1601-, AGI Exit Status: ) in new stack Oooh, now isn't that clean. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE

[asterisk-dev] [Code Review] 4020: RLS Tests: off nominal tests for lists of lists (MWI and presence)

it and that the tests that expected not to have eventlist support and had it also failed. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options

Re: [asterisk-dev] [Code Review] 4013: Alembic: Add 'outgoing' enum value to sippeers directmedia enumerator

to see if I could put 'outgoing' in the directmedia column. Downgraded, upgraded again to make sure things didn't explode. All on a postgres database Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-dev] [Code Review] 4017: chan_pjsip: Don't attempt to apply formats if there aren't any capabilities defined when creating a new channel

, now isn't that clean. 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

[asterisk-dev] [Code Review] 4013: Alembic: Add 'outgoing' enum value to sippeers directmedia enumerator

sure things didn't explode. 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

Re: [asterisk-dev] [Code Review] 4013: Alembic: Add 'outgoing' enum value to sippeers directmedia enumerator

) --- upgraded, checked to see if I could put 'outgoing' in the directmedia column. Downgraded, upgraded again to make sure things didn't explode. All on a postgres database Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided

[asterisk-dev] [Code Review] 4017: chan_pjsip: Don't attempt to apply formats if there aren't any capabilities defined when creating a new channel

/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'PJSIP/1601-' status is 'CHANUNAVAIL' -- Executing [h@default:1] NoOp(PJSIP/1601-, AGI Exit Status: ) in new stack Oooh, now isn't that clean. Thanks, Jonathan Rose

Re: [asterisk-dev] [Code Review] 3999: chan_iax2: Jitterbuffer causes crash in Asterisk 13 on account of format changes

properly. 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

Re: [asterisk-dev] [Code Review] 3990: CDRs/Dial: Fix an assertion caused by advancing a neutral state channel straight into dial pending without going through dial

transfer tests (at least ones that will run on my box... a few won't due to sipp version requirements that I really need to get around to fixing eventually) for comparison purposes. All passed exhibiting the same behavior as before as far as warning messages and such go. Thanks, Jonathan Rose

Re: [asterisk-dev] [Code Review] 3999: chan_iax2: Jitterbuffer causes crash in Asterisk 13 on account of format changes

/#review13329 --- On Sept. 16, 2014, 4:28 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3999

Re: [asterisk-dev] [Code Review] 3995: res_pjsip_endpoint_identifier_ip: Can't parse identify with match value containing CIDR

tried a few other things, such as not using a CIDR and using a hostname to verify that there wasn't any obvious deviation in behavior introduced by the patch. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-dev] [Code Review] 3990: CDRs/Dial: Fix an assertion caused by advancing a neutral state channel straight into dial pending without going through dial

--- On Sept. 18, 2014, 10:59 a.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3990

Re: [asterisk-dev] [Code Review] 3990: CDRs/Dial: Fix an assertion caused by advancing a neutral state channel straight into dial pending without going through dial

in a parallel dial. I've dialled Alice, Bob, and Charlie. I complete the dial to Alice, and publish my message. That removes the datastore. Can I be masqueraded out before I also complete the dial with Bob and Charlie? Jonathan Rose wrote: Good point. Perhaps what I

Re: [asterisk-dev] [Code Review] 3990: CDRs/Dial: Fix an assertion caused by advancing a neutral state channel straight into dial pending without going through dial

(at least ones that will run on my box... a few won't due to sipp version requirements that I really need to get around to fixing eventually) for comparison purposes. All passed exhibiting the same behavior as before as far as warning messages and such go. Thanks, Jonathan Rose

Re: [asterisk-dev] [Code Review] 3990: CDRs/Dial: Fix an assertion caused by advancing a neutral state channel straight into dial pending without going through dial

://reviewboard.asterisk.org/r/3990/#comment23819 Oops, I need a break here. - Jonathan Rose On Sept. 18, 2014, 1:21 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3990

Re: [asterisk-dev] [Code Review] 3990: CDRs/Dial: Fix an assertion caused by advancing a neutral state channel straight into dial pending without going through dial

eventually) for comparison purposes. All passed exhibiting the same behavior as before as far as warning messages and such go. Thanks, Jonathan Rose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-dev] [Code Review] 3990: CDRs/Dial: Fix an assertion caused by advancing a neutral state channel straight into dial pending without going through dial

--- On Sept. 18, 2014, 1:25 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3990/ --- (Updated Sept. 18

Re: [asterisk-dev] [Code Review] 3990: CDRs/Dial: Fix an assertion caused by advancing a neutral state channel straight into dial pending without going through dial

... a few won't due to sipp version requirements that I really need to get around to fixing eventually) for comparison purposes. All passed exhibiting the same behavior as before as far as warning messages and such go. Thanks, Jonathan Rose

Re: [asterisk-dev] [Code Review] 3990: CDRs/Dial: Fix an assertion caused by advancing a neutral state channel straight into dial pending without going through dial

that will run on my box... a few won't due to sipp version requirements that I really need to get around to fixing eventually) for comparison purposes. All passed exhibiting the same behavior as before as far as warning messages and such go. Thanks, Jonathan Rose

Re: [asterisk-dev] [Code Review] 3990: CDRs/Dial: Fix an assertion caused by advancing a neutral state channel straight into dial pending without going through dial

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3990/#review13312 --- On Sept. 11, 2014, 4:59 p.m., Jonathan Rose wrote

Re: [asterisk-dev] [Code Review] 3990: CDRs/Dial: Fix an assertion caused by advancing a neutral state channel straight into dial pending without going through dial

. 11, 2014, 4:59 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3990/ --- (Updated Sept. 11, 2014, 4:59

Re: [asterisk-dev] [Code Review] 3990: CDRs/Dial: Fix an assertion caused by advancing a neutral state channel straight into dial pending without going through dial

as warning messages and such go. 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

Re: [asterisk-dev] [Code Review] 3995: res_pjsip_endpoint_identifier_ip: Can't parse identify with match value containing CIDR

://reviewboard.asterisk.org/r/3995/#review13315 --- On Sept. 15, 2014, 1:52 p.m., Jonathan Rose wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3995

Re: [asterisk-dev] [Code Review] 3995: res_pjsip_endpoint_identifier_ip: Can't parse identify with match value containing CIDR

: 1603/sip:1603@10.24.18.13:5060;obUnknown nan Identify: 10.24.0.0/16 I tried a few other things, such as not using a CIDR and using a hostname to verify that there wasn't any obvious deviation in behavior introduced by the patch. Thanks, Jonathan Rose

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