Re: [asterisk-dev] [Code Review] 3168: PJSIP: Correct address to which ACK is sent in NAT situations

tps://reviewboard.asterisk.org/r/3168/#comment20217> You can also use pjsip_rdata_get_dlg here to retrieve a dialog, if one is present, and have this work for non-INVITE stuff as well (just in case). - Joshua Colp On Jan. 30, 2014, 7:59 p.m., opticron

Re: [asterisk-dev] [Code Review] 3168: PJSIP: Correct address to which ACK is sent in NAT situations

tps://reviewboard.asterisk.org/r/3168/#comment20219> I would actually go so far as to do a URI comparison here... cause if this happens multiple times you are going to keep growing the dialog pool usage. - Joshua Colp On Jan. 30, 2014, 8:27 p.m., opticron

Re: [asterisk-dev] [Code Review] 3168: PJSIP: Correct address to which ACK is sent in NAT situations

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3168/#review10730 --- Ship it! Ship It! - Joshua Colp On Jan. 30, 2014, 9:38 p.m

Re: [asterisk-dev] [Code Review] 3120: res_ari: Provide transfer messages and enable transfers

at messages, profited. 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] 3145: res_ari: Add event tests for blind and attended transfers

/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/bridges/attended_transfer/attended_transfer.py PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3145/diff/ Testing --- Ran test, fails using branches. Applied patch for events and it passes! Thanks, Joshua

Re: [asterisk-dev] [Code Review] 3093: res_pjsip_dialog_info: Add dialog-info+xml presence functionality.

org/r/3093/diff/ Testing --- Subscribed to an extension from a device that uses dialog-info+xml. Confirmed NOTIFY messages going out contain the expected content. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provid

[asterisk-dev] [Code Review] 3175: timing: Improve performance for most timing implementations

/ Testing --- Ran the timing test using the various timing implementations on Linux and confirmed they still work. I also placed some calls to a Playback and confirmed it works. Thanks, Joshua Colp -- _ -- Bandwidth and

[asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests

- Ran unit tests, all passed. Note: I know AO2 throws a fit and it's because a container isn't getting initialized. Getting said container initialized requires beginning the hacking apart process.

Re: [asterisk-dev] [Code Review] 3175: timing: Improve performance for most timing implementations

ed to keep my changes as minimal as possible so as to not cause regressions in that area. - Joshua --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3175/#review10747 -------

Re: [asterisk-dev] [Code Review] 3184: Create sorcery instance registry

d.asterisk.org/r/3184/#comment20259> What happens if no name is specified? Should it return NULL or maybe place a uuid in there instead? - Joshua Colp On Feb. 5, 2014, 5:08 p.m., George Joseph wrote: > > --- > This is an automat

Re: [asterisk-dev] [Code Review] 3175: timing: Improve performance for most timing implementations

: https://reviewboard.asterisk.org/r/3175/diff/ Testing --- Ran the timing test using the various timing implementations on Linux and confirmed they still work. I also placed some calls to a Playback and confirmed it works. Thanks, Joshua Colp

Re: [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate

tps://reviewboard.asterisk.org/r/3183/#comment20313> Should this actually just completely fail this? Otherwise you get a list which doesn't contain all of your JSON variables. Which, if you experienced it would cause you to scratch your head and spend time wondering what is going on. - Joshua Co

Re: [asterisk-dev] [Code Review] 3184: Create sorcery instance registry

18> This function will never get called now. The ao2 container has a reference to the sorcery instance itself. You could do something where in ast_sorcery_unref if it is the last reference it is unlinked from the instances container. - Joshua Colp On Feb. 5, 2014,

Re: [asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3180/#review10776 --- Ship it! Ship It! - Joshua Colp On Feb. 5, 2014, 9:10 p.m

Re: [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate

. /branches/12/res/ari/resource_channels.c <https://reviewboard.asterisk.org/r/3183/#comment20321> I'd throw out an error here if we got no variables. - Joshua Colp On Feb. 6, 2014, 2:51 p.m., Matt Jordan wrote: > > ---

Re: [asterisk-dev] [Code Review] 3186: AMI Security Events: document the events; add optional IEs to the events

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3186/#review10785 --- Ship it! Ship It! - Joshua Colp On Feb. 5, 2014, 10:51 p.m

Re: [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint shows allow/disallow codecs the same

ould also allow the information to still exist for cases where you do want to get it. - Joshua Colp On Feb. 3, 2014, 8:18 p.m., Scott Griepentrog wrote: > > --- > This is an automatically generated e-mail.

Re: [asterisk-dev] [Code Review] 3184: Create sorcery instance registry

isk.org/r/3184/#comment20337> Mark this as being safe so if someone looks at it they don't go "say what" - Joshua Colp On Feb. 6, 2014, 5 p.m., George Joseph wrote: > > --- > This is an automatically g

Re: [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint shows allow/disallow codecs the same - OPTION 1

> On Feb. 6, 2014, 3:52 p.m., Joshua Colp wrote: > > While this works I'm not happy with pushing this down to such a low level. > > What if in the future I want to filter something else? > > > > What I'd really like to see is something on top which

Re: [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint shows allow/disallow codecs the same - OPTION 1

> On Feb. 6, 2014, 3:52 p.m., Joshua Colp wrote: > > While this works I'm not happy with pushing this down to such a low level. > > What if in the future I want to filter something else? > > > > What I'd really like to see is something on top which

Re: [asterisk-dev] [Code Review] 3193: cli: pjsip show endpoint shows allow/disallow codecs the same - OPTION 2

get a warning, but I still dislike the idea of hidden fields. ^_^ - Joshua Colp On Feb. 6, 2014, 6:50 p.m., Scott Griepentrog wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.as

Re: [asterisk-dev] [Code Review] 3184: Create sorcery instance registry

tps://reviewboard.asterisk.org/r/3184/#comment20352> Er, do you mean you are unreffing an object that is already destroyed? If so setting the variables to NULL will prevent that. Once this clarification is provided this'll be good! - Joshua Colp On Feb. 6, 2014, 7:55 p.m., George J

Re: [asterisk-dev] [Code Review] 3184: Create sorcery instance registry

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3184/#review10804 --- Ship it! Ship It! - Joshua Colp On Feb. 6, 2014, 8:07 p.m

Re: [asterisk-dev] [Code Review] 3184: Create sorcery instance registry

> On Feb. 6, 2014, 8:21 p.m., rmudgett wrote: > > branches/12/main/sorcery.c, lines 1630-1633 > > > > > > There are two red flags here: > > 1) You are using sorcery after you have released your ref to it. > >

Re: [asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests

enerated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3178/#review10770 ------- On Feb. 7, 2014, 2:59 p.m., Joshua Colp wrote: > > --- > This is an automati

Re: [asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests

eginning the hacking apart process. 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/mailm

Re: [asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests

t: https://reviewboard.asterisk.org/r/3178/#review10821 ------- On Feb. 7, 2014, 2:59 p.m., Joshua Colp wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3178/ > ---

Re: [asterisk-dev] [Code Review] 3175: timing: Improve performance for most timing implementations

/trunk/include/asterisk/timing.h 407425 Diff: https://reviewboard.asterisk.org/r/3175/diff/ Testing --- Ran the timing test using the various timing implementations on Linux and confirmed they still work. I also placed some calls to a Playback and confirmed it works. Thanks, Joshua Colp

[asterisk-dev] [Code Review] 3199: scheduler: Remove hashtab usage.

the hashtab usage and uses the heap for finding things. Diffs - /trunk/main/sched.c 405157 Diff: https://reviewboard.asterisk.org/r/3199/diff/ Testing --- Ran scheduler unit test and also forced the scheduler to get used for Playback. Thanks, Joshua Colp

Re: [asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests

attributes, and one may be an exact match while others may be a subset. Right now the code is written to only immediately return if there's an exact match. If subsets exist then only one of them will be returned. - Joshua -

Re: [asterisk-dev] [Code Review] 3199: scheduler: Remove hashtab usage.

and also forced the scheduler to get used for Playback. 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] 3199: scheduler: Remove hashtab usage.

forced the scheduler to get used for Playback. 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] Lightweight keepalive for websockets?

o change this? It should be fine to do that. The underlying code that gets called with it accepts a const value. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- ___

Re: [asterisk-dev] AMI and Sorcery

ck in. I suppose the question really is... do we make this incompatible with the past. > Either way, this'll be good background info for why a patch to add 1 > simple dialplan function is as complex as it is. > > Bonus Question: Why are aors and auths specified as a single >

Re: [asterisk-dev] [Code Review] 3212: Makefile: fix so "make main" works without compiling deps (unless needed)

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3212/#review10866 --- Ship it! Ship It! - Joshua Colp On Feb. 12, 2014, 8:34 a.m

[asterisk-dev] [Code Review] 3234: media_formats: Initial channel driver conversion and application conversion

/app_alarmreceiver.c 408257 /team/file/media_formats-impl/apps/app_agent_pool.c 408257 Diff: https://reviewboard.asterisk.org/r/3234/diff/ Testing --- Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-dev] [Code Review] 3244: res_pjsip_sdp_rtp: Apply packetization rules on inbound SDP handling

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3244/#review10921 --- Ship it! Ship It! - Joshua Colp On Feb. 20, 2014, 10:29

[asterisk-dev] [Code Review] 3251: res_bucket_sounds: Add 'sounds' URI scheme implementation

he form of metadata. Diffs - /trunk/res/res_bucket_sounds.c PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3251/diff/ Testing --- Executed CLI commands, output is as expected. Thanks, Joshua Colp -- _ --

Re: [asterisk-dev] [Code Review] 3207: HEP: Add a Homer Encapsulation Protocol (HEP) v3 capture agent module and a packet logger for PJSIP

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3207/#review10928 --- Ship it! Ship It! - Joshua Colp On Feb. 23, 2014, 5:48 a.m

Re: [asterisk-dev] [Code Review] 3102: res_pjsip_multihomed: Add multihomed support

ct IP address for each interface went into the message. Some additional testing by others would be nice. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing li

Re: [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911

he ICE negotiation? - Joshua Colp On Feb. 24, 2014, 6:36 p.m., Jonathan Rose wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.a

Re: [asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests

container isn't getting initialized. Getting said container initialized requires beginning the hacking apart process. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk

Re: [asterisk-dev] [Code Review] 3234: media_formats: Initial channel driver conversion and application conversion

/app_amd.c 407953 /team/file/media_formats-impl/apps/app_alarmreceiver.c 407953 /team/file/media_formats-impl/apps/app_agent_pool.c 407953 Diff: https://reviewboard.asterisk.org/r/3234/diff/ Testing --- Thanks, Joshua Colp

[asterisk-dev] [Code Review] 3265: media_formats: Move app_fax over

Description --- This change moves app_fax over to the new media formats API. Diffs - /team/file/media_formats-impl/apps/app_fax.c 408263 Diff: https://reviewboard.asterisk.org/r/3265/diff/ Testing --- It compiles. Thanks, Joshua Colp

[asterisk-dev] [Code Review] 3266: media_formats: Move codecs over.

-impl/codecs/codec_adpcm.c 407954 /team/file/media_formats-impl/codecs/codec_a_mu.c 407953 Diff: https://reviewboard.asterisk.org/r/3266/diff/ Testing --- They compile, yay! Thanks, Joshua Colp -- _ -- Bandwidth and

Re: [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911

> On Feb. 24, 2014, 10:51 p.m., Joshua Colp wrote: > > /branches/11/res/res_rtp_asterisk.c, lines 554-560 > > <https://reviewboard.asterisk.org/r/3256/diff/1/?file=54416#file54416line554> > > > > Agreed, and pjnath does not provide a mechanism to do jus

Re: [asterisk-dev] [Code Review] 3245: res_pjsip_send_to_voicemail: transferring to voicemail for digium phones

_channel_unref. branches/12/res/res_pjsip_send_to_voicemail.c <https://reviewboard.asterisk.org/r/3245/#comment20572> Pfft, using RAII_VAR here is silly. - Joshua Colp On Feb. 20, 2014, 11:01 p.m., Kevin Harwell wrote: > >

Re: [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911

> On Feb. 24, 2014, 10:51 p.m., Joshua Colp wrote: > > /branches/11/res/res_rtp_asterisk.c, lines 554-560 > > <https://reviewboard.asterisk.org/r/3256/diff/1/?file=54416#file54416line554> > > > > Agreed, and pjnath does not provide a mechanism to do jus

Re: [asterisk-dev] [Code Review] 3245: res_pjsip_send_to_voicemail: transferring to voicemail for digium phones

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3245/#review10952 --- Ship it! Ship It! - Joshua Colp On Feb. 25, 2014, 5:20 p.m

Re: [asterisk-dev] [Code Review] 3267: pjsip: avoid edge case potential crash in answer()

the INVITE transaction is removed from the INVITE session. Any attempts to answer the INVITE session after this results in a crash as it requires the INVITE transaction to exist. This change explicitly locks the dialog and checks to ensure that the INVITE transaction exists before answering.

Re: [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911

> On Feb. 24, 2014, 10:51 p.m., Joshua Colp wrote: > > /branches/11/res/res_rtp_asterisk.c, lines 554-560 > > <https://reviewboard.asterisk.org/r/3256/diff/1/?file=54416#file54416line554> > > > > Agreed, and pjnath does not provide a mechanism to do jus

Re: [asterisk-dev] [Code Review] 3240: res_pjsip_exten_state: Presence for digium phones

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3240/#review10961 --- Ship it! Ship It! - Joshua Colp On Feb. 21, 2014, 5 p.m

Re: [asterisk-dev] [Code Review] 3267: pjsip: avoid edge case potential crash in answer()

tps://reviewboard.asterisk.org/r/3267/#comment20588> Throw the channel name and endpoint in here for easier debugging further down the road. - Joshua Colp On Feb. 26, 2014, 6:41 p.m., Scott Griepentrog wrote: > > --- > This is an automatica

Re: [asterisk-dev] [Code Review] 3267: pjsip: avoid edge case potential crash in answer()

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3267/#review10963 --- Ship it! Ship It! - Joshua Colp On Feb. 26, 2014, 6:52 p.m

Re: [asterisk-dev] [Code Review] 3272: func_audiohookinheritance: Check If Channel Was Specified

try to find other cases. - Joshua Colp On Feb. 27, 2014, 3:01 p.m., Michael Young wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asteris

Re: [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911

tps://reviewboard.asterisk.org/r/3256/#comment20611> There's actually a bug here you didn't cause, but can fix! If the component is RTP then rtp->s should be used, if the component is RTCP then rtp->rtcp->s should be used. Otherwise looks good! - Joshua Colp O

Re: [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911

> On Feb. 27, 2014, 6:14 p.m., Joshua Colp wrote: > > /branches/11/res/res_rtp_asterisk.c, line 1689 > > <https://reviewboard.asterisk.org/r/3256/diff/2/?file=54884#file54884line1689> > > > > There's actually a bug here you didn't cause, but can fi

Re: [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3256/#review10980 --- Ship it! Ship It! - Joshua Colp On Feb. 27, 2014, 6:41 p.m

Re: [asterisk-dev] [Code Review] 3275: res_rtp_asterisk: Fix the one way audio problems when resuming held calls with ICE

;t we always follow the pattern of putting new remote candidates into proposed and then moving them to active? - Joshua Colp On Feb. 27, 2014, 8:02 p.m., Jonathan Rose wrote: > > --- > This is an automatically generated e-mail. T

Re: [asterisk-dev] [Code Review] 3272: func_audiohookinheritance: Check If Channel Was Specified

> On Feb. 27, 2014, 4:26 p.m., Joshua Colp wrote: > > This is fine but we should probably do an audit and try to find other cases. > > Corey Farrell wrote: > If nobody is working on the audit of channel usage in dialplan functions > I can. Should I start a new JI

Re: [asterisk-dev] PJSIP transport problem

g to create two transports and bind them to the same thing - you can't do that. It's the same as trying to run two instances of Apache on port 80. Instead of binding both to 0.0.0.0 you can bind them to two different interface IP addresses if you want. -- Joshua Colp Digium, Inc. | Seni

Re: [asterisk-dev] PJSIP debug question

The res_pjsip_logger module implements the following CLI commands: pjsip set logger on pjsip set logger on pjsip set logger off You can also enable it at startup by using the "debug" option in the global section of res_pjsip.conf -- Joshua Colp Digium, Inc. | Senior Software Develo

Re: [asterisk-dev] PJSIP transport problem

options on transports control things when Asterisk is behind NAT. They specify what is local to it so SIP messages don't get rewritten, or if going outside of that - they do get rewritten to the external address. The options used when a remote endpoint is behind NAT are: force_rport=yes rtp_sym

Re: [asterisk-dev] PJSIP transport problem

times > nor send options. Is there a way to get pjsip to send out > keepalives (OPTIONS)? It's kind of a bummer to have phones > that think they are still registered, but you can't reach them. > Looked thru the literature, haven't seen anything "juicy" about > this.

Re: [asterisk-dev] [Code Review] 3266: media_formats: Move codecs over.

/codec_alaw.c 407953 /team/file/media_formats-impl/codecs/codec_adpcm.c 407954 /team/file/media_formats-impl/codecs/codec_a_mu.c 407953 Diff: https://reviewboard.asterisk.org/r/3266/diff/ Testing --- They compile, yay! Thanks, Joshua Colp

Re: [asterisk-dev] [Code Review] 3265: media_formats: Move app_fax over

t: https://reviewboard.asterisk.org/r/3265/#review10987 ------- On Feb. 25, 2014, 12:55 p.m., Joshua Colp wrote: > > --- > This is an automatically generated e-

Re: [asterisk-dev] [Code Review] 3275: res_rtp_asterisk: Fix the one way audio problems when resuming held calls with ICE

. /branches/11/res/res_rtp_asterisk.c <https://reviewboard.asterisk.org/r/3275/#comment20642> This RAII_VAR is silly. - Joshua Colp On Feb. 28, 2014, 7:04 p.m., Jonathan Rose wrote: > > --- > This is an automatically gener

Re: [asterisk-dev] [Code Review] 3275: res_rtp_asterisk: Fix the one way audio problems when resuming held calls with ICE

tps://reviewboard.asterisk.org/r/3275/#comment20658> This will crash and left_candidate will never be non-NULL. Take out that ao2_ref and this is good to go. - Joshua Colp On March 3, 2014, 5:01 p.m., Jonathan Rose wrote: > > ---

Re: [asterisk-dev] PJSIP and persistent TLS

/issues.asterisk.org/jira/browse/ASTERISK-22658 Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Band

Re: [asterisk-dev] [Code Review] 3275: res_rtp_asterisk: Fix the one way audio problems when resuming held calls with ICE

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3275/#review11028 --- Ship it! Ship It! - Joshua Colp On March 3, 2014, 6:02 p.m

Re: [asterisk-dev] PJSIP and persistent TLS

the type, IP address, and port. In the above the port for the Contact differs so it wouldn't find the existing connection. What happens if you use rewrite_contact=yes on the endpoint? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL

Re: [asterisk-dev] [Code Review] 3102: res_pjsip_multihomed: Add multihomed support

-- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3102/#review11060 --- On Feb. 23, 2014, 5:57 p.m., Joshua Colp wrote: > >

[asterisk-dev] [Code Review] 3299: res_stasis_recording: Add target URI to recording events.

iff: https://reviewboard.asterisk.org/r/3299/diff/ Testing --- Started recordings and confirmed target_uri is as expected. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-d

[asterisk-dev] [Code Review] 3303: Add test for nominal recording usage in ARI.

/recording/nominal/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3303/diff/ Testing --- Ran test, it passes. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-dev] [Code Review] 3283: pjsip_cli: Create pjsip show channel and pjsip show contact, and general cli code cleanup

> On March 6, 2014, 1:38 a.m., Matt Jordan wrote: > > branches/12/res/res_pjsip_endpoint_identifier_ip.c, lines 364-371 > > > > > > Any reason you decided to move initialization of the members of the > > struct to

Re: [asterisk-dev] [Code Review] 3283: pjsip_cli: Create pjsip show channel and pjsip show contact, and general cli code cleanup

ay be a shared object with other stuff, it may not be. Can't know. The set_extended usage has to go. Create a structure to contain additional information, put stuff in that, and pass it in. - Joshua Colp On March 3, 2014,

Re: [asterisk-dev] [Code Review] 3305: Sorcery: Be lenient when extra realtime fields are retrieved.

org/r/3305/#comment20740> Less RAII_VAR please. :P - Joshua Colp On March 5, 2014, 11:40 p.m., Mark Michelson wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.a

Re: [asterisk-dev] [Code Review] 3304: pjsip configuration: Make TOS values for transports consistent in behavior to TOS values in endpoints. Allow DSCP strings to be used for TOS configuration. Fix A

tps://reviewboard.asterisk.org/r/3304/#comment20741> Remove. - Joshua Colp On March 5, 2014, 8:19 p.m., Jonathan Rose wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.a

Re: [asterisk-dev] [Code Review] 3193: cli: pjsip show endpoint shows allow/disallow codecs the same (take 8)

o ahead and do a custom impl as well while here. - Joshua Colp On March 4, 2014, 10:56 p.m., Scott Griepentrog wrote: > > --- > This is an automatically generated e-mail. To reply, visit:

Re: [asterisk-dev] [Code Review] 3193: cli: pjsip show endpoint shows allow/disallow codecs the same (take 8)

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3193/#review11097 --- Ship it! Ship It! - Joshua Colp On March 6, 2014, 3:44 p.m

Re: [asterisk-dev] [Code Review] 3299: res_stasis_recording: Add target URI to recording events.

12/res/ari/ari_model_validators.h 409828 /branches/12/include/asterisk/stasis_app_recording.h 409828 Diff: https://reviewboard.asterisk.org/r/3299/diff/ Testing --- Started recordings and confirmed target_uri is as expected. Thanks, J

Re: [asterisk-dev] [Code Review] 3303: Add test for nominal recording usage in ARI.

/tests/rest_api/recording/nominal/recording.py PRE-CREATION /asterisk/trunk/tests/rest_api/recording/nominal/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3303/diff/ Testing --- Ran test, it passes. Thanks, Joshua Colp

Re: [asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests

I know AO2 throws a fit and it's because a container isn't getting initialized. Getting said container initialized requires beginning the hacking apart process. Thanks, Joshua Colp -- _ -- Bandwidth and Colocat

Re: [asterisk-dev] [Code Review] 3283: pjsip_cli: Create pjsip show channel and pjsip show contact, and general cli code cleanup

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3283/#review11132 --- Ship it! Ship It! - Joshua Colp On March 6, 2014, 9:51 p.m

Re: [asterisk-dev] [Code Review] 3234: media_formats: Initial channel driver conversion and application conversion

407953 /team/file/media_formats-impl/apps/app_amd.c 407953 /team/file/media_formats-impl/apps/app_alarmreceiver.c 407953 /team/file/media_formats-impl/apps/app_agent_pool.c 407953 Diff: https://reviewboard.asterisk.org/r/3234/diff/ Testing --- Thanks, Joshua Colp

Re: [asterisk-dev] [Code Review] 3266: media_formats: Move codecs over.

/codec_dahdi.c 407953 /team/file/media_formats-impl/codecs/codec_alaw.c 407953 /team/file/media_formats-impl/codecs/codec_adpcm.c 407954 /team/file/media_formats-impl/codecs/codec_a_mu.c 407953 Diff: https://reviewboard.asterisk.org/r/3266/diff/ Testing --- They compile, yay! Thanks, Joshua

Re: [asterisk-dev] [Code Review] 3265: media_formats: Move app_fax over

--- It compiles. 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

[asterisk-dev] [Code Review] 3319: media_formats: Add some additional functionality to core.

/team/group/media_formats-reviewed/include/asterisk/codec.h 410188 /team/group/media_formats-reviewed/include/asterisk/callerid.h 409286 Diff: https://reviewboard.asterisk.org/r/3319/diff/ Testing --- Thanks, Joshua Colp

[asterisk-dev] [Code Review] 3318: media_formats: Move bridges over.

Diff: https://reviewboard.asterisk.org/r/3318/diff/ Testing --- Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options

[asterisk-dev] [Code Review] 3320: media_formats: Move format modules over.

/media_formats-reviewed/formats/format_g723.c 409286 /team/group/media_formats-reviewed/formats/format_g719.c 409286 Diff: https://reviewboard.asterisk.org/r/3320/diff/ Testing --- Thanks, Joshua Colp -- _ -- Bandwidth

[asterisk-dev] [Code Review] 3322: media_formats: Move resource modules over.

/media_formats-reviewed/res/ari/resource_channels.c 409286 /team/group/media_formats-reviewed/res/ari/resource_bridges.c 409286 Diff: https://reviewboard.asterisk.org/r/3322/diff/ Testing --- Thanks, Joshua Colp

[asterisk-dev] [Code Review] 3321: media_formats: Move dialplan functions over.

/group/media_formats-reviewed/funcs/func_channel.c 409286 Diff: https://reviewboard.asterisk.org/r/3321/diff/ Testing --- Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-dev] [Code Review] 3308: tests/apps/confbridge: Add the 'triple lindy' test of marked, waitmarked, and unmarked users

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3308/#review11133 --- Ship it! Ship It! - Joshua Colp On March 6, 2014, 7:58 p.m

Re: [asterisk-dev] [Code Review] 3309: tests/apps/confbridge: Test for two waitmarked users in a conference

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3309/#review11135 --- Ship it! Ship It! - Joshua Colp On March 6, 2014, 8:02 p.m

Re: [asterisk-dev] [Code Review] 3310: tests/apps/confbridge: Add a test for waitmarked/normal user interaction

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3310/#review11136 --- Ship it! Ship It! - Joshua Colp On March 6, 2014, 8:05 p.m

Re: [asterisk-dev] [Code Review] 3311: tests/apps/agents/agent_request: Add nominal agent request test

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3311/#review11137 --- Ship it! Ship It! - Joshua Colp On March 6, 2014, 8:15 p.m

[asterisk-dev] bridge_unreal: An alternative approach to Local/Unreal channel optimization

kip any intermediary hops. (Although, since channels stick around... when would you need to use /n? Hrm...) So, thoughts? [1] http://svn.digium.com/svn/asterisk/team/file/bridge_unreal/ -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 3

Re: [asterisk-dev] bridge_unreal: An alternative approach to Local/Unreal channel optimization

e actually checks whether something like a MixMonitor is on either Local channel and won't allow it to be used. Now that I've given a diagram to show where things optimize and how it isn't inside of chan_local... what do you

Re: [asterisk-dev] bridge_unreal: An alternative approach to Local/Unreal channel optimization

n/asterisk/team/file/bridge_unreal_optimizer/ -- 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] 3102: res_pjsip_multihomed: Add multihomed support

each interface went into the message. Some additional testing by others would be nice. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or u

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