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

2014-12-05 Thread Joshua Colp
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.

2014-12-05 Thread Joshua Colp
--- 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] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard

2014-12-04 Thread Joshua Colp
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 automatically generated e-mail. To reply, visit: https

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

2014-12-03 Thread Joshua Colp
target address. - Joshua Colp On Dec. 2, 2014, 11:29 p.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4216

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

2014-12-03 Thread Joshua Colp
. 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 On Nov. 20, 2014, 9:43 p.m., Kevin Harwell wrote

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

2014-12-03 Thread Joshua Colp
--- 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

2014-12-03 Thread Joshua Colp
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 this to get spuriously triggered

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

2014-12-02 Thread Joshua Colp
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.

2014-12-02 Thread Joshua Colp
/ --- (Updated Dec. 1, 2014, 11:50 p.m.) Review request for Asterisk Developers and Joshua Colp. Repository: Asterisk Description --- Interface for the global sorcery observer /*! \brief Callback after an instance is created

Re: [asterisk-dev] [Code Review] 4212: res_pjsip_endpoint_identified_ip: Add 'show identify' and 'show identifies' cli commands

2014-12-01 Thread Joshua Colp
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4212/#review13849 --- Ship it! Ship It! - Joshua Colp On Nov. 24, 2014, 7:35 p.m

Re: [asterisk-dev] [Code Review] 4193: Stasis: allow for subscriptions to dictate message delivery on a threadpool for certain situations

2014-12-01 Thread Joshua Colp
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4193/#review13850 --- Ship it! Ship It! - Joshua Colp On Nov. 24, 2014, 6:07 p.m

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

2014-12-01 Thread Joshua Colp
if observer adding/removing is done at runtime as things progress... sadness. branches/12/main/sorcery.c https://reviewboard.asterisk.org/r/4215/#comment24334 Another magic number. - Joshua Colp On Nov. 28, 2014, 6:13 p.m., George Joseph wrote

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

2014-12-01 Thread Joshua Colp
that uses the RTP engine and res_rtp_asterisk. Something generic should be done instead, imo. - Joshua Colp On Nov. 26, 2014, 10:17 p.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https

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

2014-12-01 Thread Joshua Colp
--- 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

2014-12-01 Thread Joshua Colp
/ 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().

2014-12-01 Thread Joshua Colp
--- 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] 4204: Failure showing codecs via 'core show channeltype tech'

2014-11-24 Thread Joshua Colp
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4204/#review13844 --- Ship it! Ship It! - Joshua Colp On Nov. 24, 2014, 12:18

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

2014-11-24 Thread Joshua Colp
/res_pjsip_endpoint_identifier_ip.c https://reviewboard.asterisk.org/r/4190/#comment24315 Why does this need to happen here versus in your own wizard module? I think this also needs work for error cases to make it easier for an end-user to know what is going on when something goes wrong. - Joshua Colp On Nov. 18, 2014

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

2014-11-24 Thread Joshua Colp
On Nov. 24, 2014, 3:47 p.m., Joshua Colp wrote: branches/12/res/res_pjsip_config_wizard.c, line 295 https://reviewboard.asterisk.org/r/4190/diff/2/?file=69049#file69049line295 A question: If stuff starts falling apart and we run out of memory - how will all

Re: [asterisk-dev] [Code Review] 4185: sorcery: Make ast_sorcery_is_object_field_registered handle field names that are regexes.

2014-11-20 Thread Joshua Colp
. - Joshua Colp On Nov. 18, 2014, 9:18 p.m., George Joseph wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4185/ --- (Updated

Re: [asterisk-dev] [Code Review] 4185: sorcery: Make ast_sorcery_is_object_field_registered handle field names that are regexes.

2014-11-20 Thread Joshua Colp
/main/sorcery.c https://reviewboard.asterisk.org/r/4185/#comment24298 Regexes are uncommon and I'd prefer the normal case to be fast. Can we do a search based on key first and then fall back to a regex search? - Joshua Colp On Nov. 18, 2014, 9:18 p.m., George Joseph wrote

Re: [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete

2014-11-20 Thread Joshua Colp
On Nov. 5, 2014, 5:43 p.m., Joshua Colp wrote: Matt brought it up that this is actually a backwards incompatible change - specifically changing priority into a string from an integer. What about having label as a separate argument that is optional? If present it's treated as a label

Re: [asterisk-dev] [Code Review] 4093: Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used

2014-11-20 Thread Joshua Colp
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4093/#review13831 --- Ship it! Ship It! - Joshua Colp On Nov. 17, 2014, 2:51 a.m

Re: [asterisk-dev] [Code Review] 4185: sorcery: Make ast_sorcery_is_object_field_registered handle field names that are regexes.

2014-11-20 Thread Joshua Colp
/4185/#comment24302 This will only ever get called when a name match was not found, so you don't need to also look here. Just skip if no regex. - Joshua Colp On Nov. 20, 2014, 6:17 p.m., George Joseph wrote

Re: [asterisk-dev] [Code Review] 4193: Stasis: allow for subscriptions to dictate message delivery on a threadpool for certain situations

2014-11-20 Thread Joshua Colp
://reviewboard.asterisk.org/r/4193/#comment24305 I think this should be configurable. - Joshua Colp On Nov. 18, 2014, 6:40 p.m., Matt Jordan wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org

Re: [asterisk-dev] [Code Review] 3992: res_pjsip_sdp_rtp: Add optimistic SRTP support.

2014-11-19 Thread Joshua Colp
On Success (Encrypted) 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] 4099: Optimistic SRTP Tests.

2014-11-19 Thread Joshua Colp
happy. 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] 4196: bridge_basic: Fix features issues introduced by review 4167

2014-11-19 Thread Joshua Colp
://reviewboard.asterisk.org/r/4196/#comment24294 If you wipe the exten above then this event will always contain an empty exten in that case. - Joshua Colp On Nov. 19, 2014, 5:35 p.m., Matt Jordan wrote: --- This is an automatically generated e-mail

Re: [asterisk-dev] [Code Review] 4187: res_pjsip_refer: Ensure Refer-To is NULL terminated and parse it as a URI

2014-11-18 Thread Joshua Colp
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4187/#review13797 --- On Nov. 16, 2014, 11:36 p.m., Joshua Colp wrote

[asterisk-dev] [Code Review] 4187: res_pjsip_refer: Ensure Refer-To is NULL terminated and parse it as a URI

2014-11-16 Thread Joshua Colp
. 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] 4177: app_confbridge: Play 'leader has left' sound file even when musiconhold is enabled

2014-11-14 Thread Joshua Colp
--- On Nov. 13, 2014, 6:53 p.m., Joshua Colp wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4177

Re: [asterisk-dev] [Code Review] 4177: app_confbridge: Play 'leader has left' sound file even when musiconhold is enabled

2014-11-14 Thread Joshua Colp
/ Testing --- Joined two channels: 1 as normal, 1 as marked. Hung up the marked channel. Confirmed that the leader has left sound was played and then music on hold started. Thanks, Joshua Colp -- _ -- Bandwidth

Re: [asterisk-dev] [Code Review] 4183: res_pjsip_phoneprov_provider: bug: call ast_sorcery_apply_config so config can come from realtime.

2014-11-14 Thread Joshua Colp
://reviewboard.asterisk.org/r/4183/#comment24248 Is this really needed? ast_sorcery_open calls __ast_sorcery_open with the module name. This then calls __ast_sorcery_apply_config internally to apply configuration. Is this broken/not working? - Joshua Colp On Nov. 14, 2014, 12:34 a.m., George

Re: [asterisk-dev] [Code Review] 4093: Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used

2014-11-13 Thread Joshua Colp
On Nov. 5, 2014, 12:49 p.m., Joshua Colp wrote: /tags/12.4.0/main/rtp_engine.c, lines 2012-2018 https://reviewboard.asterisk.org/r/4093/diff/1/?file=68394#file68394line2012 This is not compliant to the way L16 is supposed to be declared within SDP. The payload name is supposed

Re: [asterisk-dev] [Code Review] 4175: Fix race condition where identical SIP requests are processed by multiple threads (Asterisk 13)

2014-11-13 Thread Joshua Colp
://reviewboard.asterisk.org/r/4175/#comment24238 This doesn't belong here. - Joshua Colp On Nov. 12, 2014, 9:51 p.m., Mark Michelson wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4175

Re: [asterisk-dev] [Code Review] 4174: Fix race condition where identical SIP requests are processed by multiple threads.

2014-11-13 Thread Joshua Colp
://reviewboard.asterisk.org/r/4174/#comment24239 Ew. - Joshua Colp On Nov. 12, 2014, 9:51 p.m., Mark Michelson wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4174

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

2014-11-13 Thread Joshua Colp
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4170/#review13743 --- Ship it! Ship It! - Joshua Colp On Nov. 11, 2014, 10:41

Re: [asterisk-dev] [Code Review] 4167: Allow mis-dialed DTMF-initiated transfers to be re-attempted.

2014-11-13 Thread Joshua Colp
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4167/#review13746 --- Ship it! Ship It! - Joshua Colp On Nov. 12, 2014, 6:32 p.m

Re: [asterisk-dev] [Code Review] 4155: PJSIP: Allow contact rewriting to fall back for reliable transports

2014-11-13 Thread Joshua Colp
://reviewboard.asterisk.org/r/4155/#comment24240 Is there a potential race condition here between transport termination and dialog termination where either may point to invalid memory? - Joshua Colp On Nov. 13, 2014, 4:53 p.m., opticron wrote

[asterisk-dev] [Code Review] 4177: app_confbridge: Play 'leader has left' sound file even when musiconhold is enabled

2014-11-13 Thread Joshua Colp
was played and then music on hold started. 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] 4162: pbx: Fix crash in off-nominal when add_priority encounters a failure.

2014-11-12 Thread Joshua Colp
. After patch: Subscribed to a hint twice - once with no spaces in it, second time with spaces in it. Running in valgrind showed no memory access issues. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http

[asterisk-dev] [Code Review] 4173: bridge: Protect bridge channel when changing state and make it smarter

2014-11-12 Thread Joshua Colp
that channels I would expect to be terminated were. 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] 4099: Optimistic SRTP Tests.

2014-11-11 Thread Joshua Colp
/tests/channels/pjsip/optimistic_srtp/mandatory_with_optimistic_offer/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4099/diff/ Testing --- Ran tests, confirmed happy. Thanks, Joshua Colp

[asterisk-dev] [Code Review] 4162: pbx: Fix crash in off-nominal when add_priority encounters a failure.

2014-11-10 Thread Joshua Colp
access issues. 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] 4162: pbx: Fix crash in off-nominal when add_priority encounters a failure.

2014-11-10 Thread Joshua Colp
- once with no spaces in it, second time with spaces in it. Running in valgrind showed no memory access issues. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing

Re: [asterisk-dev] [Code Review] 4158: rtp_engine: Fix crash when endpoints send more RTCP report info blocks then we can handle

2014-11-10 Thread Joshua Colp
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4158/#review13716 --- Ship it! Ship It! - Joshua Colp On Nov. 7, 2014, 3:55 p.m

Re: [asterisk-dev] [Code Review] 4155: PJSIP: Allow contact rewriting to fall back for reliable transports

2014-11-10 Thread Joshua Colp
/#comment24210 If my approach is viable you'd want to drop the state listener here. - Joshua Colp On Nov. 6, 2014, 3:49 p.m., opticron wrote: --- This is an automatically generated e-mail. To reply, visit: https

Re: [asterisk-dev] [Code Review] 4155: PJSIP: Allow contact rewriting to fall back for reliable transports

2014-11-10 Thread Joshua Colp
On Nov. 10, 2014, 2:24 p.m., Joshua Colp wrote: branches/12/res/res_pjsip_nat.c, line 230 https://reviewboard.asterisk.org/r/4155/diff/1/?file=68767#file68767line230 Just a question - is this already on the dialog? (Do you need to clone it?) Or can you steal it like a thief

Re: [asterisk-dev] [Code Review] 4155: PJSIP: Allow contact rewriting to fall back for reliable transports

2014-11-10 Thread Joshua Colp
On Nov. 10, 2014, 2:24 p.m., Joshua Colp wrote: branches/12/res/res_pjsip_nat.c, lines 62-77 https://reviewboard.asterisk.org/r/4155/diff/1/?file=68767#file68767line62 Instead of maintaining this in a list why not alloc the mapping from the dialog pool, pass it in as a parameter

Re: [asterisk-dev] Notes from setting up SIP+TLS/RTP+DTLS

2014-11-09 Thread Joshua Colp
consistent, just to preserve some peoples’ hair? Maybe. All of the DTLS configuration occurs in common logic across the code base. Provided it's fed what is expected then it's happy. It sounds like overall we just need better documentation of this. Cheers, -- Joshua Colp Digium, Inc. | Senior

Re: [asterisk-dev] Notes from setting up SIP+TLS/RTP+DTLS

2014-11-09 Thread Joshua Colp
for SDES you need to protect the SIP signaling, or else someone will know your encryption key. In the case of DTLS since it's negotiated outside of the signaling it doesn't matter as much. The most they could see is the fingerprint of the certificate on each side. Cheers, -- Joshua Colp

Re: [asterisk-dev] [Code Review] 4157: bridge_native_rtp: Fix T.38 directmedia fax test by always asking the remote peers to update themselves on native bridge stop

2014-11-07 Thread Joshua Colp
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4157/#review13706 --- Ship it! Ship It! - Joshua Colp On Nov. 7, 2014, 3:50 p.m

Re: [asterisk-dev] [Code Review] 4146: res_pjsip: If an endpoint is associated with the dialog place it on the messag early

2014-11-06 Thread Joshua Colp
no endpoint would be present on the 200 OK response to an outgoing re-INVITE. With the patch an endpoint is present on the 200 OK response. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-dev] Asterisk 14 - Remote URI Playback

2014-11-06 Thread Joshua Colp
it comes from. This being said there's nothing in the sorcery core acting as a cache or place for this stuff - that's up to the wizards. As well in bucket what wizard is used is determined based on the URI scheme - and there can be only one impl for each scheme. Cheers, -- Joshua Colp Digium

Re: [asterisk-dev] func_jitterbuffer handling of masquerades

2014-11-06 Thread Joshua Colp
at that point? Probably not. Just as a slight note: ConfBridge does not remove the jitterbuffer when the channel leaves it. Same for denoise. _ -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com

Re: [asterisk-dev] [Code Review] 4148: res_pjsip_multihomed: Provide logging during startup for an indication of what is going on

2014-11-05 Thread Joshua Colp
/4148/diff/ Testing --- Ran and confirmed messages output. 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] 4149: main/file.c: fix possible extra ast_module_unref to format modules

2014-11-05 Thread Joshua Colp
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4149/#review13683 --- Ship it! Ship It! - Joshua Colp On Nov. 4, 2014, 11:11 p.m

Re: [asterisk-dev] [Code Review] 4093: Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used

2014-11-05 Thread Joshua Colp
? - Joshua Colp On Oct. 31, 2014, 1:32 a.m., Frankie Chin wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4093

Re: [asterisk-dev] [Code Review] 3603: func_jitterbuffer: fix audio failure caused by certain masquerade's

2014-11-05 Thread Joshua Colp
don't remember it so I don't think so. Before accepting the behavior you've done I'd really like us to talk about it there amongst everyone. I really think it's the reverse of what a user would expect to happen when a masquerade happens. - Joshua Colp On Oct. 31, 2014, 12:06 a.m., Corey

Re: [asterisk-dev] [Code Review] 4053: res_pjsip_history: A debugging module for busy systems

2014-11-05 Thread Joshua Colp
On Oct. 16, 2014, 9:44 p.m., Mark Michelson wrote: /trunk/res/res_pjsip_history.c, lines 44-47 https://reviewboard.asterisk.org/r/4053/diff/1/?file=67862#file67862line44 I wonder if the code could be simplified by instead of saving rdata and tdata, saving a pjsip_msg structure

Re: [asterisk-dev] [Code Review] 4053: res_pjsip_history: A debugging module for busy systems

2014-11-05 Thread Joshua Colp
://reviewboard.asterisk.org/r/4053/#comment24181 This can be allocated without a lock. - Joshua Colp On Oct. 8, 2014, 1:55 p.m., Matt Jordan wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4053

Re: [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete

2014-11-05 Thread Joshua Colp
for a developer using this to debug what is going on. - Joshua Colp On Nov. 2, 2014, 1:25 p.m., greenfieldtech wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4101

Re: [asterisk-dev] [Code Review] 3867: [chan_sip] Default DTLS settings to use if peer misses own settings

2014-11-05 Thread Joshua Colp
for anyone else to chime in). I also apologize for how long this has taken. We strive for better but don't always hit where we want to be. - Joshua Colp On Aug. 3, 2014, 8:57 a.m., Michael K. wrote: --- This is an automatically generated e

Re: [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete

2014-11-05 Thread Joshua Colp
and also makes it a bit more explicit from a developer side of what they want. - Joshua Colp On Nov. 5, 2014, 2:16 p.m., greenfieldtech wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org

Re: [asterisk-dev] [Code Review] 4102: testsuite: add secure websocket test

2014-11-05 Thread Joshua Colp
server is available, and thus WSS is available. - Joshua Colp On Oct. 21, 2014, 11:06 p.m., Scott Griepentrog wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4102

Re: [asterisk-dev] [Code Review] 3603: func_jitterbuffer: fix audio failure caused by certain masquerade's

2014-11-05 Thread Joshua Colp
On Nov. 5, 2014, 12:55 p.m., Joshua Colp wrote: Was the behavior for this ever brought up on the -dev list? I don't remember it so I don't think so. Before accepting the behavior you've done I'd really like us to talk about it there amongst everyone. I really think it's the reverse

Re: [asterisk-dev] func_jitterbuffer handling of masquerades

2014-11-05 Thread Joshua Colp
, and allowing some method to control it confuses people. That's my feelings about this. What do others think? -- 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] 4139: stun: fix size of attribute string to match rfc

2014-11-05 Thread Joshua Colp
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4139/#review13696 --- Ship it! Ship It! - Joshua Colp On Oct. 31, 2014, 8:29 p.m

Re: [asterisk-dev] [Code Review] 2964: res_pjsip_outbound_registration: Add virtual line support for automatic inbound matching

2014-11-04 Thread Joshua Colp
from it, confirmed matched using line parameter. 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] 4099: Optimistic SRTP Tests.

2014-11-04 Thread Joshua Colp
--- On Oct. 21, 2014, 1:49 p.m., Joshua Colp wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4099/ --- (Updated Oct. 21, 2014

[asterisk-dev] [Code Review] 4148: res_pjsip_multihomed: Provide logging during startup for an indication of what is going on

2014-11-04 Thread Joshua Colp
once done tells you what IP addresses it has used. Diffs - /branches/12/res/res_pjsip_multihomed.c 427199 Diff: https://reviewboard.asterisk.org/r/4148/diff/ Testing --- Ran and confirmed messages output. Thanks, Joshua Colp

Re: [asterisk-dev] [Code Review] 4150: res_hep: fix major leak that occurs when config is missing or enabled=no

2014-11-04 Thread Joshua Colp
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4150/#review13677 --- Ship it! Ship It! - Joshua Colp On Nov. 5, 2014, 1:05 a.m

Re: [asterisk-dev] [Code Review] 4103: chan_pjsip: Add 'moh_passthrough' option for passing through musiconhold requests.

2014-11-03 Thread Joshua Colp
/ Testing --- Enabled option. Placed call to a remote server. Put call on hold and off hold. Confirmed re-INVITEs were sent with proper state. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-dev] [asterisk-commits] file: trunk r427112 - in /trunk: channels/ channels/pjsip/ configs/samples/ c...

2014-11-03 Thread Joshua Colp
Matthew Jordan wrote: Should have caught this in the review, but do you mind updating the CHANGES file for this as well? :-) Done! But hey - I remembered alembic! -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out

Re: [asterisk-dev] Fwd: [asterisk-commits] mjordan: branch 12 r426995 - /branches/12/res/res_stasis.c

2014-11-03 Thread Joshua Colp
. AO2_GLOBAL_OBJ_STATIC uses a read/write lock and the time it's accessed should be relatively small enough that meh - penalty shouldn't be that bad for this sorta stuff. That was my personal concern when I saw you mention that. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis

[asterisk-dev] [Code Review] 4146: res_pjsip: If an endpoint is associated with the dialog place it on the messag early

2014-11-03 Thread Joshua Colp
. 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] 3992: res_pjsip_sdp_rtp: Add optimistic SRTP support.

2014-10-29 Thread Joshua Colp
--- On Oct. 21, 2014, 1:36 p.m., Joshua Colp wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3992

Re: [asterisk-dev] SRTP in chan_motif.c

2014-10-27 Thread Joshua Colp
dwal...@fifo99.com wrote: On Mon, Oct 27, 2014 at 07:01:10PM -0300, Joshua Colp wrote: As there is no crypto support written it is not possible to negotiate it. Support would have to be added. All 3 variants are implemented. Non-Documented Google, Google, and XEP. So Google Voice's protocol

Re: [asterisk-dev] SRTP in chan_motif.c

2014-10-27 Thread Joshua Colp
dwal...@fifo99.com wrote: On Mon, Oct 27, 2014 at 07:24:27PM -0300, Joshua Colp wrote: dwal...@fifo99.com wrote: On Mon, Oct 27, 2014 at 07:01:10PM -0300, Joshua Colp wrote: As there is no crypto support written it is not possible to negotiate it. Support would have to be added. All 3

Re: [asterisk-dev] asterisk-11.13.1 nits

2014-10-24 Thread Joshua Colp
is 20995d419dace207f828a3a8463c22e1 if that helps. 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 and Colocation

Re: [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

2014-10-22 Thread Joshua Colp
up and wanted to talk about - it's not something that has been decided 'nor does anyone know what the future entails. Any further discussions will naturally occur on the mailing list and in fact some things have explicit action items to bring them up on here. Cheers, -- Joshua Colp Digium, Inc

[asterisk-dev] [Code Review] 4103: chan_pjsip: Add 'moh_passthrough' option for passing through musiconhold requests.

2014-10-22 Thread Joshua Colp
. 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] 4105: codec_dahdi: Fix segfault on load.

2014-10-22 Thread Joshua Colp
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4105/#review13589 --- Ship it! Ship It! - Joshua Colp On Oct. 22, 2014, 7:13 p.m

Re: [asterisk-dev] [Code Review] 3992: res_pjsip_sdp_rtp: Add optimistic SRTP support.

2014-10-21 Thread Joshua Colp
Optimistic Off Success (Encrypted) Mandatory Optimistic On Success (Encrypted) Optional Optimistic Off Success (Encrypted) Optional Optimistic On Success (Encrypted) Thanks, Joshua Colp

Re: [asterisk-dev] [Code Review] 3992: res_pjsip_sdp_rtp: Add optimistic SRTP support.

2014-10-21 Thread Joshua Colp
On Success (Encrypted) 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] [Code Review] 4099: Optimistic SRTP Tests.

2014-10-21 Thread Joshua Colp
/mandatory_with_optimistic_offer/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4099/diff/ Testing --- Ran tests, confirmed happy. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-dev] [Code Review] 4073: res_pjsip: Add 'user_eq_phone' option for placing 'user=phone' parameter in request URI if user is number.

2014-10-17 Thread Joshua Colp
425395 Diff: https://reviewboard.asterisk.org/r/4073/diff/ Testing --- Sent outgoing calls with various users (numbers, number+letters, letters) and confirmed that user=phone was set when it should be. Thanks, Joshua Colp

Re: [asterisk-dev] [Code Review] 2964: res_pjsip_outbound_registration: Add virtual line support for automatic inbound matching

2014-10-17 Thread Joshua Colp
--- On Oct. 10, 2014, 1:18 p.m., Joshua Colp wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/2964

Re: [asterisk-dev] [Code Review] 3992: res_pjsip_sdp_rtp: Add optimistic SRTP support.

2014-10-17 Thread Joshua Colp
, 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] 4084: res_pjsip_keepalive: Add keepalive module for connection-oriented transports.

2014-10-17 Thread Joshua Colp
, 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] 4084: res_pjsip_keepalive: Add keepalive module for connection-oriented transports.

2014-10-16 Thread Joshua Colp
, visit: https://reviewboard.asterisk.org/r/4084/#review13527 --- On Oct. 15, 2014, 6:08 p.m., Joshua Colp wrote: --- This is an automatically generated e-mail. To reply, visit

Re: [asterisk-dev] [Code Review] 4084: res_pjsip_keepalive: Add keepalive module for connection-oriented transports.

2014-10-16 Thread Joshua Colp
/pjsip.conf.sample 425689 Diff: https://reviewboard.asterisk.org/r/4084/diff/ Testing --- Configured keepalives and used wireshark to verify that at the specific interval the message went out. Thanks, Joshua Colp

Re: [asterisk-dev] [Code Review] 4053: res_pjsip_history: A debugging module for busy systems

2014-10-16 Thread Joshua Colp
://reviewboard.asterisk.org/r/4053/#comment24069 I'm not a fan of these scoped mutexes being here. Despite the enabled check being fast you've still got a contention point on every message. Is it just protecting the vector or enabled also? - Joshua Colp On Oct. 8, 2014, 1:55 p.m., Matt Jordan wrote

Re: [asterisk-dev] [Code Review] 4063: res_pjsip_session/res_pjsip_sdp_rtp: Fix a variety of situations where Asterisk would incorrectly reject offers

2014-10-16 Thread Joshua Colp
https://reviewboard.asterisk.org/r/4063/#comment24070 Yay! /branches/13/res/res_pjsip_sdp_rtp.c https://reviewboard.asterisk.org/r/4063/#comment24071 FYI I killed this in optimistic. - Joshua Colp On Oct. 14, 2014, 6:54 p.m., Matt Jordan wrote

Re: [asterisk-dev] [Code Review] 4079: testsuite: Update Offer/Answer PJSIP Tests

2014-10-16 Thread Joshua Colp
. - Joshua Colp On Oct. 14, 2014, 7:07 p.m., Matt Jordan wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4079/ --- (Updated Oct

Re: [asterisk-dev] [Code Review] 4084: res_pjsip_keepalive: Add keepalive module for connection-oriented transports.

2014-10-16 Thread Joshua Colp
/res_pjsip.h 425690 /trunk/configs/samples/pjsip.conf.sample 425690 Diff: https://reviewboard.asterisk.org/r/4084/diff/ Testing --- Configured keepalives and used wireshark to verify that at the specific interval the message went out. Thanks, Joshua Colp

Re: [asterisk-dev] [Code Review] 4089: config: Fix infinite loop when using ast_category_browse and ast_variable_retrieve together.

2014-10-16 Thread Joshua Colp
://reviewboard.asterisk.org/r/4089/#comment24075 I don't think this test goes far enough. I'd make each category have different values for the variables and ensure that each retrieve in each category gets the right thing. - Joshua Colp On Oct. 16, 2014, 3:37 p.m., George Joseph wrote

Re: [asterisk-dev] [Code Review] 4089: config: Fix infinite loop when using ast_category_browse and ast_variable_retrieve together.

2014-10-16 Thread Joshua Colp
/4089/#comment24076 In the case of retrieving a variable I don't think this will retrieve the correct information. Since the search is done from the root it'll find the first occurrence. My suggestion for test change will confirm it. - Joshua Colp On Oct. 16, 2014, 3:37 p.m., George Joseph

Re: [asterisk-dev] [Code Review] 4089: config: Fix infinite loop when using ast_category_browse and ast_variable_retrieve together.

2014-10-16 Thread Joshua Colp
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4089/#review13548 --- Ship it! Ship It! - Joshua Colp On Oct. 16, 2014, 5:18 p.m

Re: [asterisk-dev] [Code Review] 4053: res_pjsip_history: A debugging module for busy systems

2014-10-16 Thread Joshua Colp
On Oct. 16, 2014, 11:32 a.m., Joshua Colp wrote: /trunk/res/res_pjsip_history.c, line 87 https://reviewboard.asterisk.org/r/4053/diff/1/?file=67862#file67862line87 I'm not a fan of these scoped mutexes being here. Despite the enabled check being fast you've still got a contention

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