Re: [asterisk-dev] [Code Review] 3679: WebRTC: Add SHA-256 support, change DTLS-SRTP negotiation, add finer grain control of things.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3679/#review12388 --- I do not feel qualified to add a ship-it flag, but can confirm

Re: [asterisk-dev] [Code Review] 3686: WebRTC: Add SHA-256 support to chan_pjsip and add option to make it answer using the offer media transport.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3686/#review12389 --- /branches/12/res/res_pjsip.c

Re: [asterisk-dev] [Code Review] 3607: [app_queue] Add the optional ability to load queue rules from realtime.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3607/ --- (Updated June 30, 2014, 1:26 p.m.) Review request for Asterisk Developers.

Re: [asterisk-dev] Proposed removal of deprecated modules in Asterisk 13 (cdr_sqlite, chan_gtalk, chan_jingle, res_jabber, chan_h323, app_readfile, app_dahdibarge, app_setcallerid, app_saycountpl)

On Sat, Jun 28, 2014 at 5:50 PM, Matthew Jordan wrote: > The following modules are currently deprecated in trunk. This e-mail > is a proposal to remove them from trunk in preparation for Asterisk > 13. All of these modules have been deprecated for at least two release > cycles (the one they were d

Re: [asterisk-dev] Proposed removal of deprecated modules in Asterisk 13 (cdr_sqlite, chan_gtalk, chan_jingle, res_jabber, chan_h323, app_readfile, app_dahdibarge, app_setcallerid, app_saycountpl)

On 29-06-14 00:50, Matthew Jordan wrote: * app_readfile/app_dahdibarge/app_setcallerid/app_saycountpl - deprecated in Asterisk 1.8. While less important than the other previously listed modules, having been deprecated for 2 LTS's, it is probably time for these module to go. May I also suggest r

Re: [asterisk-dev] [Code Review] 3686: WebRTC: Add SHA-256 support to chan_pjsip and add option to make it answer using the offer media transport.

> On June 30, 2014, 1:12 p.m., Matt Jordan wrote: > > /branches/12/res/res_pjsip_sdp_rtp.c, lines 541-545 > > > > > > That seems like an odd setting return here. > > > > Why wouldn't this return AST_SIP_MED

Re: [asterisk-dev] Proposed removal of deprecated modules in Asterisk 13 (cdr_sqlite, chan_gtalk, chan_jingle, res_jabber, chan_h323, app_readfile, app_dahdibarge, app_setcallerid, app_saycountpl)

On Mon, Jun 30, 2014 at 9:24 AM, Walter Doekes wrote: > On 29-06-14 00:50, Matthew Jordan wrote: > >> * app_readfile/app_dahdibarge/app_setcallerid/app_saycountpl - >> deprecated in Asterisk 1.8. While less important than the other >> previously listed modules, having been deprecated for 2 LTS's,

Re: [asterisk-dev] [Code Review] 3686: WebRTC: Add SHA-256 support to chan_pjsip and add option to make it answer using the offer media transport.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3686/ --- (Updated June 30, 2014, 2:31 p.m.) Review request for Asterisk Developers.

Re: [asterisk-dev] [Code Review] 3686: WebRTC: Add SHA-256 support to chan_pjsip and add option to make it answer using the offer media transport.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3686/#review12391 --- /branches/12/contrib/ast-db-manage/config/versions/51f8cb66540

Re: [asterisk-dev] [Code Review] 3686: WebRTC: Add SHA-256 support to chan_pjsip and add option to make it answer using the offer media transport.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3686/ --- (Updated June 30, 2014, 3:20 p.m.) Review request for Asterisk Developers.

Re: [asterisk-dev] [Code Review] 3686: WebRTC: Add SHA-256 support to chan_pjsip and add option to make it answer using the offer media transport.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3686/#review12392 --- Ship it! Ship It! - Matt Jordan On June 30, 2014, 10:20 a.m

[asterisk-dev] Regarding packet2packet bridging

Dear concern, I want to configure packet2packet bridging in asterisk. How could I do this any of the tutorial or instructions will help ? I found the setting the canreinvite=yes will do the stuff but it is not working I am using asterisk 12.3 version I am very new to asterisk please help me i

Re: [asterisk-dev] Regarding packet2packet bridging

On Mon, Jun 30, 2014 at 12:29 PM, Sameer Rathod wrote: > Dear concern, > > > I want to configure packet2packet bridging in asterisk. > How could I do this any of the tutorial or instructions will help ? > > I found the setting the canreinvite=yes will do the stuff but it is not > working > > I a

[asterisk-dev] [Code Review] 3690: CEL: Fix incorrect/missing extra field information

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3690/ --- Review request for Asterisk Developers. Repository: Asterisk Description

Re: [asterisk-dev] [Code Review] 3687: media format improvements: Update packetization handling; improve rtp_engine's ast_rtp_codecs handling

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3687/#review12393 --- I have not yet reviewed rtp_engine.c - that one will take a bit

Re: [asterisk-dev] [Code Review] 3686: WebRTC: Add SHA-256 support to chan_pjsip and add option to make it answer using the offer media transport.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3686/ --- (Updated June 30, 2014, 7:47 p.m.) Status -- This change has been mar

Re: [asterisk-dev] [Code Review] 3679: WebRTC: Add SHA-256 support, change DTLS-SRTP negotiation, add finer grain control of things.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3679/ --- (Updated June 30, 2014, 2:42 p.m.) Status -- This change has been mar

Re: [asterisk-dev] [Code Review] 3690: CEL: Fix incorrect/missing extra field information

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3690/ --- (Updated June 30, 2014, 2:57 p.m.) Review request for Asterisk Developers.

Re: [asterisk-dev] [Code Review] 3690: CEL: Fix incorrect/missing extra field information

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3690/ --- (Updated June 30, 2014, 3:04 p.m.) Review request for Asterisk Developers

Re: [asterisk-dev] [Code Review] 3687: media format improvements: Update packetization handling; improve rtp_engine's ast_rtp_codecs handling

> On June 30, 2014, 2:27 p.m., Corey Farrell wrote: > > /team/group/media_formats-reviewed-trunk/channels/chan_sip.c, lines > > 10088-10090 > > > > > > Why switch these to be undefined values? Compilation issue.

Re: [asterisk-dev] [Code Review] 3687: media format improvements: Update packetization handling; improve rtp_engine's ast_rtp_codecs handling

> On June 30, 2014, 3:27 p.m., Corey Farrell wrote: > > /team/group/media_formats-reviewed-trunk/channels/chan_sip.c, lines > > 10088-10090 > > > > > > Why switch these to be undefined values? > > Matt Jordan wr

Re: [asterisk-dev] [Code Review] 3687: media format improvements: Update packetization handling; improve rtp_engine's ast_rtp_codecs handling

> On June 30, 2014, 2:27 p.m., Corey Farrell wrote: > > /team/group/media_formats-reviewed-trunk/channels/chan_sip.c, lines > > 10088-10090 > > > > > > Why switch these to be undefined values? > > Matt Jordan wr

Re: [asterisk-dev] [Code Review] 3687: media format improvements: Update packetization handling; improve rtp_engine's ast_rtp_codecs handling

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3687/ --- (Updated June 30, 2014, 5:14 p.m.) Review request for Asterisk Developers.

Re: [asterisk-dev] [Code Review] 3690: CEL: Fix incorrect/missing extra field information

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3690/#review12397 --- branches/12/main/causes.c

Re: [asterisk-dev] [Code Review] 3601: accountcode: Slightly change accountcode propagation.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3601/ --- (Updated June 30, 2014, 6:24 p.m.) Review request for Asterisk Developers.

Re: [asterisk-dev] [Code Review] 3690: CEL: Fix incorrect/missing extra field information

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3690/#review12398 --- branches/12/include/asterisk/causes.h

Re: [asterisk-dev] [Code Review] 3687: media format improvements: Update packetization handling; improve rtp_engine's ast_rtp_codecs handling

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3687/ --- (Updated June 30, 2014, 7:02 p.m.) Review request for Asterisk Developers.

Re: [asterisk-dev] [Code Review] 3687: media format improvements: Update packetization handling; improve rtp_engine's ast_rtp_codecs handling

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3687/#review12399 --- /team/group/media_formats-reviewed-trunk/formats/format_h263.c

Re: [asterisk-dev] [Code Review] 3690: CEL: Fix incorrect/missing extra field information

> On June 30, 2014, 5:54 p.m., Matt Jordan wrote: > > branches/12/main/causes.c, lines 19-24 > > > > > > Since this is specific to sip, I'd place it in something that calls > > that out. Maybe sip_causes? I figured

Re: [asterisk-dev] [Code Review] 3690: CEL: Fix incorrect/missing extra field information

> On June 30, 2014, 6:38 p.m., Corey Farrell wrote: > > branches/12/res/res_pjsip_session.c, lines 1808-1810 > > > > > > What about space for the NULL terminator? The space for the null terminator is allocated in

[asterisk-dev] [Code Review] 3691: HTTP: Add persistent connection support.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3691/ --- Review request for Asterisk Developers. Bugs: ASTERISK-23552 https://i