Re: [asterisk-dev] [Code Review] 4211: Speed up loopback switches by avoiding unneeded lookups

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4211/#review13853 --- Ship it! Ship It! - Joshua Colp On Nov. 24, 2014, 1:15 a.m

[asterisk-dev] [Code Review] 4218: res_pjsip_refer: Remove framehook when transfer is marked as completed as a result of joining a bridge

/ Testing --- An environment was set up with multiple phones and a blind transfer was performed. Before the patch the bridge after completion of the transfer would be of type simple_bridge. With the patch the bridge after completion would be of type native_rtp. Thanks, Joshua Colp

Re: [asterisk-dev] [Code Review] 4217: config: Create ast_variable_find_in_list().

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4217/#review13855 --- Ship it! Ship It! - Joshua Colp On Dec. 1, 2014, 4:19 p.m

Re: [asterisk-dev] [Code Review] 4218: res_pjsip_refer: Remove framehook when transfer is marked as completed as a result of joining a bridge

would be of type simple_bridge. With the patch the bridge after completion would be of type native_rtp. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list

Re: [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.

generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4215/ > --- > > (Updated Dec. 1, 2014, 11:50 p.m.) > > > Review request for Asterisk Developers and Joshua Colp. > > > Repository: Aster

Re: [asterisk-dev] [Code Review] 4216: res_pjsip: Direct Media calls within private network sometimes get one way audio

ource address" and the other be the "requested target address". - Joshua Colp On Dec. 2, 2014, 11:29 p.m., Kevin Harwell wrote: > > --- > This is an automatically generated e-mail

Re: [asterisk-dev] [Code Review] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard

7;s possible for this to get spuriously triggered. As for the time limit - can things be in a state where that is safe to do? Some logging would also be useful so people know what is going on if it waits a bit. - Joshua Colp

Re: [asterisk-dev] [Code Review] 4221: test framework: Fix race condition between AMI topic and Test Suite topic raising of AMI events

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4221/#review13876 --- Ship it! Ship It! - Joshua Colp On Dec. 3, 2014, 1:53 p.m

Re: [asterisk-dev] [Code Review] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard

> On Dec. 3, 2014, 4:48 p.m., Joshua Colp wrote: > > branches/13/res/res_pjsip_outbound_publish.c, lines 1268-1269 > > <https://reviewboard.asterisk.org/r/4178/diff/2/?file=69153#file69153line1268> > > > > You need to be in a loop here. It's possible for

Re: [asterisk-dev] [Code Review] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard

rd.asterisk.org/r/4178/#comment24397> Per my race condition comment I think you may want to just end up doing this at load time. - Joshua Colp On Dec. 4, 2014, 6:13 p.m., Kevin Harwell wrote: > > --- > This is an automati

Re: [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4215/#review13903 --- Ship it! Ship It! - Joshua Colp On Dec. 5, 2014, 4:01 p.m

[asterisk-dev] [Code Review] 4244: ari: Add originate with linkedid test.

/ Testing --- Ran test, it passes! 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

[asterisk-dev] [Code Review] 4243: ari: Add the ability to specify a linkedId when originating calls.

that the resulting CEL events contain the provided linked ID. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-dev] [Code Review] 4244: ari: Add originate with linkedid test.

/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4244/diff/ Testing --- Ran test, it passes! Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev

Re: [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify a linkedId when originating calls.

ID of the originator. 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/lis

Re: [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify a linkedId when originating calls.

------ On Dec. 5, 2014, 9:10 p.m., Joshua Colp wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4243/ >

Re: [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify an originator when originating calls.

hanks, 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] 4243: ari: Add the ability to specify a linkedId when originating calls.

n all existing ARI origination tests and confirmed they still pass. Performed calls manually and examined the resulting channels and CEL log to ensure they contain the linked ID of the originator. Thanks, Joshua

Re: [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify an originator when originating calls.

n all existing ARI origination tests and confirmed they still pass. Performed calls manually and examined the resulting channels and CEL log to ensure they contain the linked ID of the originator. Thanks, Joshua

Re: [asterisk-dev] [Code Review] 4244: ari: Add originate with originator test.

--- Ran test, it passes! 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

Re: [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios

/12/res/res_pjsip_config_wizard.c <https://reviewboard.asterisk.org/r/4190/#comment24422> This doesn't clean up the object_type_wizards vector or the objects stored in the wizard. - Joshua Colp On Dec. 3, 2014, 11:42 p.m., G

Re: [asterisk-dev] [Code Review] 4245: ARI/AMI: Include language in standard channel snapshot output

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4245/#review13912 --- Ship it! Ship It! - Joshua Colp On Dec. 5, 2014, 11:12 p.m

Re: [asterisk-dev] [Code Review] 4170: testsuite: Delete bridges on completion for a bunch of rest_api tests

? - Joshua Colp On Nov. 11, 2014, 10:41 p.m., Corey Farrell wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asteris

Re: [asterisk-dev] [Code Review] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard

/res_pjsip_outbound_publish.c <https://reviewboard.asterisk.org/r/4178/#comment24439> I'd opt for verbose messages for these. - Joshua Colp On Dec. 4, 2014, 10:26 p.m., Kevin Harwell wrote: > > --- > This is an automatically generated e-m

Re: [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios

> On Dec. 8, 2014, 12:52 p.m., Joshua Colp wrote: > > branches/12/res/res_pjsip_config_wizard.c, lines 261-283 > > <https://reviewboard.asterisk.org/r/4190/diff/3/?file=69442#file69442line261> > > > > While forgiving this would yield an incomplete variable

Re: [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios

> On Dec. 8, 2014, 12:52 p.m., Joshua Colp wrote: > > branches/12/res/res_pjsip_config_wizard.c, line 746 > > <https://reviewboard.asterisk.org/r/4190/diff/3/?file=69442#file69442line746> > > > > Is this is successful and last_config exists you may want to j

Re: [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify an originator when originating calls.

hanks, 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] 4244: ari: Add originate with originator test.

--- Ran test, it passes! 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

Re: [asterisk-dev] Rules for binding with regards to RTP

ntial to cause issues, e.g. with NAT or symmetric RTP? So, yes. In an ICE scenario the code will provide candidates for all local addresses. If you are bound to only one then all the others will fail. This should be taken into account - but it ultimately (potentially) lessens the chance of a better ro

Re: [asterisk-dev] Rules for binding with regards to RTP

Mark Michelson wrote: On 12/09/2014 01:42 PM, Joshua Colp wrote: Mark Michelson wrote: So, yes. In an ICE scenario the code will provide candidates for all local addresses. If you are bound to only one then all the others will fail. This should be taken into account - but it ultimately

[asterisk-dev] [Code Review] 4248: res_pjsip_session: Queue BYE if an outstanding INVITE transaction exists

--- Manually tested and confirmed the behavior executes. Ran directmedia transfer tests and confirmed they were written to take this behavior into account. After fixing the behavior they were broken and are fixed in another review. Thanks, Joshua Colp

[asterisk-dev] [Code Review] 4249: PJSIP Blind Transfer Direct Media Tests: Update to expect sane behavior

--- Ran tests against Asterisk using patch from 4248 and they are once again happy. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update

[asterisk-dev] [Code Review] 4255: Testsuite: Add test for SDP offer/answer during hold with a declined stream

/asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/incoming/off-nominal/multiple-media-stream/audio-video/hold-declined/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4255/diff/ Testing --- Ran test, it lives! Thanks, Joshua Colp

[asterisk-dev] [Code Review] 4254: res_pjsip_session: Don't allow declined media streams to fail SDP negotiation on re-INVITE

continued to work as expected. 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

Re: [asterisk-dev] [Code Review] 4254: res_pjsip_session: Don't allow declined media streams to fail SDP negotiation on re-INVITE

existing SDP offer/answer tests to make sure they continued to work as expected. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update

Re: [asterisk-dev] [Code Review] 4250: Sorcery: Log when a stale configuration remains in use

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4250/#review13941 --- Ship it! Ship It! - Joshua Colp On Dec. 10, 2014, 10:21

Re: [asterisk-dev] [Code Review] 4252: PJSIP: Allow use of 'inactive' stream types for hold

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4252/#review13942 --- Ship it! Ship It! - Joshua Colp On Dec. 11, 2014, 4:43 p.m

Re: [asterisk-dev] [Code Review] 4255: Testsuite: Add test for SDP offer/answer during hold with a declined stream

! 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] 4230: add capath support to res_pjsip

tps://reviewboard.asterisk.org/r/4230/#comment24479> As this requires a pjproject change this needs a configure check so it can be optional instead of a hard dependency. As it is the build would just fail on an older install. - Joshua Colp On Dec. 4, 2014, 10:09 p.m., cloos

Re: [asterisk-dev] [Code Review] 4254: res_pjsip_session: Don't allow declined media streams to fail SDP negotiation on re-INVITE

/ Testing --- Ran test. Before patch it failed, with patch it worked. I also ran the existing SDP offer/answer tests to make sure they continued to work as expected. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation

Re: [asterisk-dev] [Code Review] 4255: Testsuite: Add test for SDP offer/answer during hold with a declined stream

/4255/diff/ Testing --- Ran test, it lives! Thanks, Joshua Colp -- _ -- 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] 4248: res_pjsip_session: Queue BYE if an outstanding INVITE transaction exists

. After fixing the behavior they were broken and are fixed in another review. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update

Re: [asterisk-dev] [Code Review] 4249: PJSIP Blind Transfer Direct Media Tests: Update to expect sane behavior

/pjsip/transfers/blind_transfer/callee_direct_media/sipp/alice.xml 6082 Diff: https://reviewboard.asterisk.org/r/4249/diff/ Testing --- Ran tests against Asterisk using patch from 4248 and they are once again happy. Thanks, Joshua Colp

[asterisk-dev] [Code Review] 4260: media: Fix crash when determining sample count of a frame during shutdown

generally hard to reproduce for me. 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

Re: [asterisk-dev] [Code Review] 4257: chan_pjsip: Race condition between channel answer and bridge setup when using direct media

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4257/#review13955 --- Ship it! I also +1 ship it. - Joshua Colp On Dec. 11, 2014

Re: [asterisk-dev] [Code Review] 4260: media: Fix crash when determining sample count of a frame during shutdown

/ Testing --- Ran some normal calls to ensure no regression. Also did some restarts/shutdowns but due to the raceyness of this it is generally hard to reproduce for me. Thanks, Joshua Colp -- _ -- Bandwidth and

Re: [asterisk-dev] Problem developing asterisk module with libcurl

ady curl functionality in Asterisk - does it not fit your needs? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _

Re: [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4190/#review13960 --- Ship it! Ship It! - Joshua Colp On Dec. 10, 2014, 11:21

Re: [asterisk-dev] AVPF support in asterisk

? There is no implemented functionality for AVPF. The RTP stack doesn't generate or consume any. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.ast

Re: [asterisk-dev] [Code Review] 4269: res_pjsip_config_wizard: fix test breakage

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4269/#review13966 --- Ship it! Ship It! - Joshua Colp On Dec. 16, 2014, 5:49 p.m

Re: [asterisk-dev] [Code Review] 4271: ARI: Allow interoperation of ASYNCGOTO and channels originated to Stasis()

in this case is the bridge in a state where we can't do an explicit goto? Is this check wrong? - Joshua Colp On Dec. 16, 2014, 8:41 p.m., opticron wrote: > > --- > This is an automatically generated e-mail. To reply,

Re: [asterisk-dev] [Code Review] 4277: Ensure that CHANNEL(pjsip, secure) gives expected value.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4277/#review13981 --- Ship it! Ship It! - Joshua Colp On Dec. 17, 2014, 3:36 p.m

Re: [asterisk-dev] [Code Review] 4268: During a dual redirect, prevent a race condition that may cause one of the redirected channels to be hung up.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4268/#review13982 --- Ship it! Ship It! - Joshua Colp On Dec. 16, 2014, 4:24 p.m

Re: [asterisk-dev] [Code Review] 4271: ARI: Allow interoperation of ASYNCGOTO and channels originated to Stasis()

tps://reviewboard.asterisk.org/r/4271/#comment24509> It also *may* have been broken out by an ASYNCGOTO softhangup flag. It may not have (ala continue). - Joshua Colp On Dec. 17, 2014, 5:54 p.m., opticron wrote: > > --- > This is an automatica

Re: [asterisk-dev] [Code Review] 4288: res_pjsip_phoneprovi_provider: Fix reload

tps://reviewboard.asterisk.org/r/4288/#comment24534> Why are you using this instead of a retrieve_by_fields and AST_RETRIEVE_FLAG_MULTIPLE | AST_RETRIEVE_FLAG_ALL? - Joshua Colp On Dec. 21, 2014, 10:52 p.m., George Joseph wrote: > > --

Re: [asterisk-dev] [Code Review] 4288: res_pjsip_phoneprovi_provider: Fix reload

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4288/#review14024 --- Ship it! Ship It! - Joshua Colp On Dec. 22, 2014, midnight

[asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions

en provided (which is done with PJSIP_DIAL_CONTACTS) then only contact would return a value. This would allow the pre-dial handler to query for additional information about the contact being dialed in order to do things. Thoughts? -- Joshua Colp Digium, Inc. | Senior Software Devel

Re: [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions

e or not is to have separate endpoints. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Col

Re: [asterisk-dev] [Code Review] 4286: rtp_engine: avoid payload types above 127

g/r/4286/#comment24538> Yeah, put this in the 96-127 rang instead. - Joshua Colp On Dec. 19, 2014, 8:24 p.m., Scott Griepentrog wrote: > > --- > This is an automatically generated e-mail. To reply,

Re: [asterisk-dev] [Code Review] 4285: Bug fixes for ARI Originate/Continue with label support (Continuation of /r/4101)

tps://reviewboard.asterisk.org/r/4285/#comment24540> I don't think this is the correct message to put here. They didn't request priority 0, it just so happens that what they provided yielded us to having a priority of 0. I could see something being confused if they saw this. - Joshua Colp

Re: [asterisk-dev] [Code Review] 4293: testsuite: Add a test for PJSIP keep alive packets for connection oriented transports

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4293/#review14030 --- Ship it! Ship It! - Joshua Colp On Dec. 23, 2014, 4:19 a.m

Re: [asterisk-dev] [Code Review] 4294: testsuite: Add a test for user_eq_phone setting in PJSIP

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4294/#review14031 --- Ship it! Ship It! - Joshua Colp On Dec. 23, 2014, 12:46

Re: [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions

nels Then, conceivably, the dialplan could be used to determine which contacts on the AoR map to what Contacts were provided by the channels. If there isn't a one-to-one mapping, it at least becomes the domain of the person building the system to resolve the discrepancy, and not something that

Re: [asterisk-dev] [Code Review] 4300: bridge_native_rtp: Change local/remote message from debug/2 to verb/4

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4300/#review14046 --- Ship it! Ship It! - Joshua Colp On Dec. 29, 2014, 11:09

Re: [asterisk-dev] [Code Review] 4296: PJSIP: Fix bugs and improve documentation of remote attended transfers

> On Dec. 24, 2014, 9:46 p.m., Kevin Harwell wrote: > > /branches/13/res/res_pjsip_refer.c, lines 604-605 > > > > > > I don't think pjsip places a null terminator on the filled in string. > > May need to get the r

Re: [asterisk-dev] [Code Review] 4297: config: Add option to NOT preserve the effective context when changing a template.

s a poor variable for this because it may end up being other things in the future - Joshua Colp On Dec. 29, 2014, 1:22 a.m., George Joseph wrote: > > --- > This is an automatically generated e-mail. To reply, vi

Re: [asterisk-dev] [Code Review] 4298: Testsuite: Tests for Manager/Config/NoPreserveEffectiveContext

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4298/#review14052 --- Ship it! Ship It! - Joshua Colp On Dec. 29, 2014, 1:29 a.m

Re: [asterisk-dev] [Code Review] 4298: Testsuite: Tests for Manager/Config/NoPreserveEffectiveContext

/no_preserve_effective_context/test-config.yaml <https://reviewboard.asterisk.org/r/4298/#comment24568> now - Joshua Colp On Dec. 29, 2014, 1:29 a.m., George Joseph wrote: > > --- > This is an automatically generated e-mail. To reply,

Re: [asterisk-dev] [Code Review] 4283: Testsuite: Dual channel redirect tests

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4283/#review14050 --- Ship it! Ship It! - Joshua Colp On Dec. 24, 2014, 7:01 p.m

Re: [asterisk-dev] [Code Review] 4301: res_pjsip_outbound_registration: Add 'pjsip send register' command and update behavior of 'send unregister'

an unregister this is just a plain ol' register. branches/13/res/res_pjsip_outbound_registration.c <https://reviewboard.asterisk.org/r/4301/#comment24571> I'm concerned that the behavior that used to happen was explicitly documented, it wasn't overlooked. We're now

Re: [asterisk-dev] [Code Review] 4286: rtp_engine: avoid payload types above 127

> On Dec. 23, 2014, 1:41 p.m., Joshua Colp wrote: > > /trunk/main/rtp_engine.c, line 2193 > > <https://reviewboard.asterisk.org/r/4286/diff/1/?file=69966#file69966line2193> > > > > Yeah, put this in the 96-127 rang instead. > > Scott Griep

Re: [asterisk-dev] [Code Review] 4286: rtp_engine: avoid payload types above 127

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4286/#review14061 --- Ship it! Ship It! - Joshua Colp On Dec. 31, 2014, 6:06 p.m

[asterisk-dev] [Code Review] 4308: pjsip: Add PJSIP_AOR and PJSIP_CONTACT dialplan functions

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

[asterisk-dev] [Code Review] 4310: pjsip :Add tests for PJSIP_AOR and PJSIP_CONTACT dialplan functions

then failed. 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

Re: [asterisk-dev] [Code Review] 4310: pjsip :Add tests for PJSIP_AOR and PJSIP_CONTACT dialplan functions

- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4310/#review14066 ------- On Jan. 2, 2015, 6:05 p.m., Joshua Colp wrote: > > -

Re: [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions

same Contact exists across different AORs then all channels will show up on all of them (even if different endpoint/AOR). This may be right, this may be wrong. It depends. It also means we would have to make the rules for matching of the Contact between the one on the AOR and the one in the sessio

Re: [asterisk-dev] [Code Review] 4308: pjsip: Add PJSIP_AOR and PJSIP_CONTACT dialplan functions

/pjsip/dialplan_functions.c 430178 Diff: https://reviewboard.asterisk.org/r/4308/diff/ Testing --- Ran tests, they're happy. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digita

Re: [asterisk-dev] [Code Review] 4285: Bug fixes for ARI Originate/Continue with label support (Continuation of /r/4101)

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4285/#review14098 --- Ship it! Ship It! - Joshua Colp On Jan. 6, 2015, 10:21 p.m

Re: [asterisk-dev] [Code Review] 4284: Testsuite: Update ARI test for continuation and add new ARI test for origination to dialplan location

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4284/#review14099 --- Ship it! Nice naming of things. - Joshua Colp On Dec. 19

Re: [asterisk-dev] [Code Review] 4297: config: Add option to NOT preserve the effective context when changing a template.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4297/#review14100 --- Ship it! Ship It! - Joshua Colp On Dec. 30, 2014, 9:45 p.m

Re: [asterisk-dev] [Code Review] 4272: Testsuite: Verify that bridged channels originated into Stasis and AMI Redirect interoperate properly

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4272/#review14101 --- Ship it! Ship It! - Joshua Colp On Dec. 29, 2014, 4:17 p.m

Re: [asterisk-dev] [Code Review] 4304: res_pjsip_outbound_registration: Fix several reload issues.

utbound registration '%s' again branches/13/res/res_pjsip_outbound_registration.c <https://reviewboard.asterisk.org/r/4304/#comment24602> Calling sip_outbound_registration_apply here is overkill. You should only need to push a sip_outbound_registration_perform task using the exi

Re: [asterisk-dev] [Code Review] 4310: pjsip :Add tests for PJSIP_AOR and PJSIP_CONTACT dialplan functions

--- Ran... so far away. Er I mean they passed. Sabotaged some things and confirmed they then failed. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing

Re: [asterisk-dev] [Code Review] 4313: func_config: Add ability to retrieve specific occurrence of a variable

debug message should be extended to include the index. It may even be useful to show the highest index it got to. - Joshua Colp On Jan. 2, 2015, 6:39 p.m., George Joseph wrote: > > --- > This is an automatical

Re: [asterisk-dev] [Code Review] 4306: res_pjsip_mwi: Change "MWI Subscription failed" message from warning to verb/4.

other review. - Joshua Colp On Jan. 1, 2015, 11:29 p.m., George Joseph wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.as

Re: [asterisk-dev] [Code Review] 4307: res_pjsip_exten_state: Change 'does not exist' warning to verb/4

other review. - Joshua Colp On Jan. 2, 2015, 2:21 a.m., George Joseph wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.as

Re: [asterisk-dev] [Code Review] 4313: func_config: Add ability to retrieve specific occurrence of a variable

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4313/#review14109 --- Ship it! Ship It! - Joshua Colp On Jan. 7, 2015, 4:29 p.m

Re: [asterisk-dev] [Code Review] 4304: res_pjsip_outbound_registration: Fix several reload issues.

tps://reviewboard.asterisk.org/r/4304/#comment24608> Did you test this? It was a "should be fine" statement. - Joshua Colp On Jan. 7, 2015, 3:52 p.m., George Joseph wrote: > > --- > This is an automatically generate

Re: [asterisk-dev] [Code Review] 4306: res_pjsip_mwi: Change "MWI Subscription failed" message from warning to verb/4.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4306/#review14113 --- Ship it! Ship It! - Joshua Colp On Jan. 7, 2015, 6:09 p.m

Re: [asterisk-dev] [Code Review] 4307: res_pjsip_exten_state: Change 'does not exist' warning to verb/4

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4307/#review14114 --- Ship it! Ship It! - Joshua Colp On Jan. 7, 2015, 6:07 p.m

Re: [asterisk-dev] [Code Review] 4304: res_pjsip_outbound_registration: Fix several reload issues.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4304/#review14121 --- Ship it! Ship It! - Joshua Colp On Jan. 7, 2015, 3:52 p.m

Re: [asterisk-dev] [Code Review] 4325: PJSIP: Prevent slow graceful shutdown when outbound publications have not started

tps://reviewboard.asterisk.org/r/4325/#comment24628> explicit_publish_destroy? - Joshua Colp On Jan. 9, 2015, 12:07 a.m., Mark Michelson wrote: > > --- > This is an automatically generated e-mail. To reply,

Re: [asterisk-dev] [Code Review] 4311: res_pjsip: make it unloadable

> On Jan. 8, 2015, 9:13 p.m., Mark Michelson wrote: > > branches/13/res/res_pjsip.c, lines 43-45 > > > > > > What caused these dependencies to be added? > > Kevin Harwell wrote: > pjsip_options uses sorcery_me

Re: [asterisk-dev] [Code Review] 4311: res_pjsip: make it unloadable

Harwell wrote: > pjsip_options uses sorcery_memory with regards to qualify/contact_status > and contacts themselves are stored in the astdb (location.c) > > Joshua Colp wrote: > I think res_sorcery_memory is fine being a dependency because it is > actually required by stuff intern

Re: [asterisk-dev] [Code Review] 4311: res_pjsip: make it unloadable

Harwell wrote: > pjsip_options uses sorcery_memory with regards to qualify/contact_status > and contacts themselves are stored in the astdb (location.c) > > Joshua Colp wrote: > I think res_sorcery_memory is fine being a dependency because it is > actually required by stuff intern

Re: [asterisk-dev] [Code Review] 4325: PJSIP: Prevent slow graceful shutdown when outbound publications have not started

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4325/#review14145 --- Ship it! Ship It! - Joshua Colp On Jan. 9, 2015, 3:45 p.m

[asterisk-dev] [Code Review] 4328: res_pjsip: Document transport selection process

://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

Re: [asterisk-dev] [Code Review] 4328: res_pjsip: Document transport selection process

orgot a URL there. Fixed. - Joshua --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4328/#review14156 --- On Jan. 1

[asterisk-dev] [Code Review] 4329: chan_pjsip: Add configure check for presence of 'pjsip_get_dest_info' function.

.asterisk.org/r/4329/diff/ Testing --- Ran configure check, confirmed it found the support. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSU

Re: [asterisk-dev] [Code Review] 4328: res_pjsip: Document transport selection process

ail. 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 gener

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