[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: http

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.

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/listinfo

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

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

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

, 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.

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 Colp

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

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 Colp

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

this get placed in the wizard order? Is it at the start, at the end, etc. branches/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

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.asterisk.org/r/4170

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-mail. To reply, visit

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 list, causing the resulting object

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

, 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

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 route. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445

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: snip more snips 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

[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

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 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

the 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

://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 wrote

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

of this it is 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

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

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 -- _ -- Bandwidth

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.asterisk.org

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, visit: https

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()

/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 automatically generated e-mail

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

://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

handler to query for additional information about the contact being dialed in order to do things. Thoughts? -- 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] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions

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 Colocation Provided

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

/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, visit: https://reviewboard.asterisk.org/r

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

://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 On Dec. 19, 2014

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

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 Asterisk itself has to figure out. I'd be down with this. Cheers, -- Joshua Colp

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 https://reviewboard.asterisk.org/r/4296/diff/1/?file=70013#file70013line604 I don't think pjsip places a null terminator on the filled in string. May need to get the result of the

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

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, visit: https://reviewboard.asterisk.org/r/4297

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, visit: https

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'

. 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 changing it - why was it the previous way in the first place? - Joshua Colp

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 Griepentrog wrote: Wouldn't that cause ast_rtp_engine_load_format

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] 4310: pjsip :Add tests for PJSIP_AOR and PJSIP_CONTACT dialplan functions

they 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

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

, 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] 4310: pjsip :Add tests for PJSIP_AOR and PJSIP_CONTACT dialplan functions

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

Re: [asterisk-dev] [Code Review] 4356: realtime: Updates fail to work due to update fields being passed over for lookup fields

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4356/#review14274 --- Ship it! Ship It! - Joshua Colp On Jan. 20, 2015, 3:10 a.m

Re: [asterisk-dev] [Code Review] 4345: Use SIPS Contact headers as prescribed by RFC 3261 (res_pjsip)

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4345/#review14273 --- Ship it! Ship It! - Joshua Colp On Jan. 15, 2015, 8:07 p.m

Re: [asterisk-dev] [Code Review] 4363: res_pjsip: make it unloadable (take 2)

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4363/#review14275 --- Ship it! Ship It! - Joshua Colp On Jan. 21, 2015, 4:30 p.m

[asterisk-dev] [Code Review] 4375: res_parking: Unsubscribe and join when unloading.

a few times, it unloaded. 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] 4383: res_pjsip_config_wizard: Add ability to auto-create hints.

(and by extension people subscribed to it using something like SIP)? This also opens up a window where someone may try to use the configured extension and it won't be available. - Joshua Colp On Jan. 30, 2015, 6:16 p.m., George Joseph wrote

Re: [asterisk-dev] [Code Review] 4370: RAII_VAR: nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality. TAKE 2

/4370/#comment24928 If the compiler being used is clang I think this should be required here and not just a warning when you actually try to compile. If the support doesn't exist it's just not going to work and having configure succeed is just putting off the inevitable failure. - Joshua

Re: [asterisk-dev] [Code Review] 4400: Add no_answer to ARI hangup causes

. - Joshua Colp On Feb. 3, 2015, 4:30 p.m., Ben Merrills wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4400/ --- (Updated

Re: [asterisk-dev] [Code Review] 4365: Adding AMQP backend for CDR and CEL

Ditto. - Joshua Colp On Feb. 6, 2015, 1:13 a.m., David Lee wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4365

[asterisk-dev] [Code Review] 4338: res_pjsip_outbound_registration: Fix race condition with executing pjsip show registrations and reloading.

://reviewboard.asterisk.org/r/4338/diff/ Testing --- Ran show registrations and reloads over and over, still happy. Without patch this would sometimes crash. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-dev] [Code Review] 4331: res_pjsip_multihomed: Account for when the transport on a message is already correct

transport on the message is already correct it is not altered. Diffs - /branches/13/res/res_pjsip_multihomed.c 430545 Diff: https://reviewboard.asterisk.org/r/4331/diff/ Testing --- Ran test and confirmed it failed without patch, passed with patch. Thanks, Joshua Colp

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

://wiki.asterisk.org/wiki/display/~jcolp/Transport+Selection Diffs - Diff: https://reviewboard.asterisk.org/r/4328/diff/ Testing --- I opened the wiki page. It opened. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation

Re: [asterisk-dev] [Code Review] 4331: res_pjsip / res_pjsip_multihomed: Use the correct transport and addressing information on UAS sessions

and makes sense. Diffs (updated) - /branches/13/res/res_pjsip_multihomed.c 430683 /branches/13/res/res_pjsip.c 430683 Diff: https://reviewboard.asterisk.org/r/4331/diff/ Testing --- Ran test and confirmed it failed without patch, passed with patch. Thanks, Joshua Colp

Re: [asterisk-dev] [Code Review] 4335: res_pjsip_multihomed: Test to ensure correct message contents when multiple transports exist

/incoming_call_on_second_transport/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4335/diff/ Testing --- Ran it. It passed. Sabotaged it. It failed. Rolled back fix. It failed. Thanks, Joshua Colp

Re: [asterisk-dev] [Code Review] 4331: res_pjsip / res_pjsip_multihomed: Use the correct transport and addressing information on UAS sessions

/res/res_pjsip_multihomed.c 430683 /branches/13/res/res_pjsip.c 430683 Diff: https://reviewboard.asterisk.org/r/4331/diff/ Testing --- Ran test and confirmed it failed without patch, passed with patch. Thanks, Joshua Colp

Re: [asterisk-dev] [Code Review] 4345: Use SIPS Contact headers as prescribed by RFC 3261 (res_pjsip)

this: Why can we not construct a proper URI from the beginning, why is a module required to alter it afterwards? - Joshua Colp On Jan. 15, 2015, 8:07 p.m., Mark Michelson wrote: --- This is an automatically generated e-mail. To reply

Re: [asterisk-dev] [Code Review] 4316: ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis dialplan application to another system; improve and fix PJSIP's transfer ability

/res_pjsip_transport_websocket.c https://reviewboard.asterisk.org/r/4316/#comment24685 Etc. - Joshua Colp On Jan. 19, 2015, 3:16 a.m., Matt Jordan wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4316

Re: [asterisk-dev] [Code Review] 4350: AMI: Add documentation for the missing Cdr/CEL events.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4350/#review14225 --- Ship it! Ship It! - Joshua Colp On Jan. 17, 2015, 5:32 p.m

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

. Joshua Colp wrote: That's not the transport being spoken of. It literally means a SIP URI such as sip:1000@127.0.0.1;transport=tcp Joshua Colp wrote: I've tweaked things some, added some endpoints, and referenced the applicable section that will be used depending

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

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4328/#review14194 --- On Jan. 12, 2015, 1:33 p.m., Joshua Colp wrote

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

. As a side note, SRV may change your port address. Joshua Colp wrote: I've tweaked the page slightly to include this condition but have not gone further. Specifically because this page is for how a local transport is selected based on the existing code and not the process of target

Re: [asterisk-dev] [Code Review] 4338: res_pjsip_outbound_registration: Fix race condition with executing pjsip show registrations and reloading.

, 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] 4429: bridge_softmix: G.729 codec license held

, and the lifetime of the helper entry. As it is it's very... confusing... - Joshua Colp On Feb. 16, 2015, 6:30 p.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https

Re: [asterisk-dev] [Code Review] 4428: Allow graceful shutdown to unload modules that register bucket scheme's or codec's.

/4428/#comment25012 Why the !mod addition? Something trigger it? - Joshua Colp On Feb. 16, 2015, 6:58 p.m., Corey Farrell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r

Re: [asterisk-dev] [Code Review] 4427: Fix crash when AOC-E message is received after the channel has been destroyed.

On Feb. 16, 2015, 6:50 p.m., Matt Jordan wrote: /branches/12/main/aoc.c, lines 1871-1880 https://reviewboard.asterisk.org/r/4427/diff/1/?file=71450#file71450line1871 I don't think this is the right way to fix the problem. This is really just working around the

Re: [asterisk-dev] [Code Review] 4428: Allow graceful shutdown to unload modules that register bucket scheme's or codec's.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4428/#review14493 --- Ship it! Ship It! - Joshua Colp On Feb. 16, 2015, 6:58 p.m

Re: [asterisk-dev] [Code Review] 4375: res_parking: Unsubscribe and join when unloading.

- /branches/13/res/parking/parking_manager.c 431112 Diff: https://reviewboard.asterisk.org/r/4375/diff/ Testing --- Unloaded module a few times, it unloaded. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation

Re: [asterisk-dev] [Code Review] 4371: Update res_format_attr_opus res_format_attr_silk to new media formats architecture

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4371/#review14307 --- Ship it! Ship It! - Joshua Colp On Jan. 26, 2015, 11:38

[asterisk-dev] [Code Review] 4378: bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during direct media.

: https://reviewboard.asterisk.org/r/4378/diff/ Testing --- Tried various scenarios including attended transfers and multiple Asterisk instances in the path. Previously media would go via the wrong route or not at all. With patch reinvites occur as expected. Thanks, Joshua Colp

Re: [asterisk-dev] [Code Review] 4384: res_pjsip_outbound_publish: eventually crashes when no response is ever received

On Jan. 29, 2015, 4:14 p.m., Joshua Colp wrote: branches/13/res/res_pjsip_outbound_publish.c, line 927 https://reviewboard.asterisk.org/r/4384/diff/2/?file=71172#file71172line927 So... This may actually cause publish state to be reflected incorrectly on the remote

Re: [asterisk-dev] [Code Review] 4387: res_pjsip_exten_state: Reduce log clutter... change a WARNING to a VERBOSE/2.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4387/#review14353 --- Ship it! Ship It! - Joshua Colp On Jan. 29, 2015, 1:10 a.m

Re: [asterisk-dev] [Code Review] 4384: res_pjsip_outbound_publish: eventually crashes when no response is ever received

://reviewboard.asterisk.org/r/4384/#comment24866 What does a user see when this happens? Any indication? - Joshua Colp On Jan. 28, 2015, 11:44 p.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https

Re: [asterisk-dev] [Code Review] 4384: res_pjsip_outbound_publish: eventually crashes when no response is ever received

then it will be out of order. As well - wouldn't this cause to the queue to potentially grow uncontrollably? - Joshua Colp On Jan. 29, 2015, 3:55 p.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply

Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1

, we could split the concept of the endpoint name from the auth user, which would then allow the endpoints to be named 107 (for example) and the auth user to be her MAC address. Joshua Colp wrote: I think in practice this would just cause problems. Not all devices allow those two things

Re: [asterisk-dev] [Code Review] 4394: chan_pjsip: Port over attribute passthrough tests and add test for sprop-parameter-sets.

/ Testing --- Ran tests. They ran. 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] 4392: res_format_attr_h264: Fix crash when using sprop-parameter-sets.

. 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] 4378: bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during direct media.

would go via the wrong route or not at all. With patch reinvites occur as expected. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE

Re: [asterisk-dev] [Code Review] 4378: bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during direct media.

. Previously media would go via the wrong route or not at all. With patch reinvites occur as expected. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list

Re: [asterisk-dev] [Code Review] 4378: bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during direct media.

with the technology join/leave callbacks. Otherwise, what is the point of the join/leave technology callbacks if the technology is going to look at the bridge channel member list at inappropriate times? Joshua Colp wrote: That flag can be unset and reset appropriately. The bridge

Re: [asterisk-dev] [Code Review] 4378: bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during direct media.

://reviewboard.asterisk.org/r/4378/#review14308 --- On Jan. 27, 2015, 12:06 p.m., Joshua Colp wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4378

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