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

to the > 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&d=DwMCaQ&c=q3cDpHe1hF8lXU5EFjNM_A&r=50uagQBTpQAKCx3KjAwJcM

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

e blind parkAndAnnounce") > same => > 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 in

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 metho

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 a

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 actua

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 chann

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 functionality is imple

[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 matche

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 that box. > > Is there any re

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 > wrote: > >> Hello, >> I am looking for an advice for having a bug fixed. I am talking about >> ASTERISK-254

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 > asterisk.. You guys have

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 1.4.2

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 ide

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

screw 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: >

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> on behalf of Sean Bright

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

ested 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] 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 with

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

Jonathan Rose has uploaded a new patch set (#3). 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 with

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

achines 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 UNSUBS

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

d never occur. I'm probably being paranoid. - Jonathan --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4577/#review15075 --- On Ap

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

patch 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 P

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

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

[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

[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

[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

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

oject: testsuite Gerrit-Branch: master Gerrit-Owner: Jonathan Rose Gerrit-Reviewer: Ashley Sanders Gerrit-Reviewer: Matt Jordan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing li

[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...

https://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 Gerrit-Reviewer: Ashley Sanders G

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

view, 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 Gerrit-Rev

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

tps://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,

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

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 updat

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

diohooks instead of framehooks actually... but it's the same concept. - Jonathan --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4577/#review15034 ------

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

. To reply, visit: https://reviewboard.asterisk.org/r/4577/#review15034 --- On April 2, 2015, 2:08 p.m., Jonathan Rose wrote: > > --- > This is an automatica

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

Jonathan Rose has uploaded a new change for review. https://gerrit.asterisk.org/22 Change subject: Add a test for PJSIP t38 with authentication based on normal t38 test .. Add a test for PJSIP t38 with authentication based

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

. 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

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

he 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 -- ast

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] 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/#review14915 --- Ship it! Ship It! - Jonathan Rose On March 27, 2015, 11:39

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

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!

tps://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,

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

tps://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.a

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

or now. - Jonathan --- 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.,

[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 To

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

ps://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 occu

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

> On March 16, 2015, 8:57 a.m., Matt Jordan wrote: > > /branches/13/configs/basic-pbx/extensions.conf, lines 36-42 > > > > > > Does this really need to be a separate context? > > > > I'm all for having conte

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

tps://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

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

or 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, Jon

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

oad Module: 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, Jon

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

oad Module: 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 visi

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

using that tech 3 - Use 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

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

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

avior didn't differ from 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

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

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

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 or update option

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

tched the status 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/l

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

k.org/r/4109/diff/ Testing --- 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 sc

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

vices are UNAVAILABLE and/or UNREGISTERED Ringing - All devices are RINGING InUse&Ringing - All devices are RINGING and one or more devices are INUSE Hold - All devices are ONHOLD. InUse&Hold - All devices are ONHO

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

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4085/#review13529 --- On Oct. 15, 2014, 2:01 p.m., Jonathan Rose wrote: >

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

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

Hold - All devices are ONHOLD. InUse&Hold - 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 Col

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

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

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

thout seeing any assertions of this type 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 -- _

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

hanks, 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

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

[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] 4071: scheduler: Fix a bug introduced by adding a delete flag to scheduled tasks

hout 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://www.api-digital.com -- asterisk-dev mailing list To

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

m 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://lists.digium.com/mailman/listinfo/asterisk-dev

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

eters 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, Jon

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.asteris

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.asteris

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

org/r/4013/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

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

|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|slin|)) to (nothing) [Sep 22 16:01:59] WARNING[12441][C-]: app_dial.c:2431 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/

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

ive (g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|slin|)) to (nothing) [Sep 22 16:01:59] WARNING[12441][C-]: app_dial.c:2431 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/

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

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 -- _ -- Bandwidth

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

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

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

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 upda

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

to (nothing) [Sep 22 16:01:59] WARNING[12441][C-]: app_dial.c:2431 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1

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 -- _ -- B

[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

Unable to create channel of type 'PJSIP' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'PJSIP/1601-' status is 'CHANUNAVAIL' -- Executing [h@default:1] NoOp("PJSIP/1601-

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

g (updated) --- 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 -- _ -- Bandwidt

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

graded again to make 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:/

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

al end and dial begin events... not sure if this is a problem, but I might be able to fix it just by updating the old chan, not sure what status code to use though). Ran against normal attended transfer, feature attended transfers, and blind transfers with no noticeable effect. Ran against testsu

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

uld get started 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/li

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

e if this is a problem, but I might be able to fix it just by updating the old chan, not sure what status code to use though). Ran against normal attended transfer, feature attended transfers, and blind transfers with no noticeable effect. Ran against testsuite blonde transfer tests, som

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

is a problem, but I might be able to fix it just by updating the old chan, not sure what status code to use though). Ran against normal attended transfer, feature attended transfers, and blind transfers with no noticeable effect. Ran against testsuite blonde transfer tests, some attended tra

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/#review13345 --- On Sept. 18, 2014, 1:25 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

ATION","1410472131.1","" And dial events reported by AMI: http://pastebin.com/tWuwL7xa (note that there is a mismatch between the number of dial end and dial begin events... not sure if this is a problem, but I might be able to fix it jus

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

tps://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://reviewbo

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

end and dial begin events... not sure if this is a problem, but I might be able to fix it just by updating the old chan, not sure what status code to use though). Ran against normal attended transfer, feature attended transfers, and blind transfers with no noticeable effect. Ran against testsu

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

irst dial > > completion message. > > > > Say I'm 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 ou

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

an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3990/#review13339 --- On Sept. 18, 2014, 10:59 a.m., Jonathan Rose wrote: > > --- > T

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

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 -- ___

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

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

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 a problem, but I might be able to fix it just by updating the old chan, not sure what status code to use though). Ran against normal attended transfer, feature attended transfers, and blind transfers with no noticeable effect. Ran against testsuite blonde transfer tests, some attended tra

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