> On Dec. 4, 2013, 9:02 p.m., Mark Michelson wrote: > > /branches/12/channels/pjsip/dialplan_functions.c, lines 316-327 > > <https://reviewboard.asterisk.org/r/3038/diff/1/?file=48949#file48949line316> > > > > These descriptions are inaccurate when used on outgoing channels. > > Matt Jordan wrote: > Doh. > > How about: > > target_uri: The request URI of the INVITE request associated with the > creation of this channel. > local_uri - The URI in the To header of the INVITE request associated > with the creation of this channel. > remote_uri - The URI in the From header of the INVITE request associated > with the creation of this channel.
local_uri and remote_uri are still off. For outbound calls, the local_uri will be the URI in the From header and the remote_uri will be the URI in the To header. > On Dec. 4, 2013, 9:02 p.m., Mark Michelson wrote: > > /branches/12/channels/pjsip/dialplan_functions.c, lines 338-345 > > <https://reviewboard.asterisk.org/r/3038/diff/1/?file=48949#file48949line338> > > > > These descriptions are inaccurate when used on outbound channels. > > Matt Jordan wrote: > Not sure this is much better: > > local_addr: The full IP address and port number that received/transmitted > the INVITE request associated with the creation of this channel. > remote_addr: The full IP address and port number that sent/was the target > of the INVITE request associated with the creation of this channel. That's better, though it is a touch awkward. If you're not too worried about brevity you can just use two separate sentences for each. Example text for local_addr: "On inbound calls, the full IP address and port number that the INVITE request was received on. On outbound calls, the full IP address and port number that the INVITE request was transmitted to." - Mark ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3038/#review10299 ----------------------------------------------------------- On Dec. 10, 2013, 3:51 a.m., Matt Jordan wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3038/ > ----------------------------------------------------------- > > (Updated Dec. 10, 2013, 3:51 a.m.) > > > Review request for Asterisk Developers. > > > Repository: Asterisk > > > Description > ------- > > This patch adds CHANNEL function support to chan_pjsip. Since things were > getting a bit large, all dialplan functions that were in chan_pjsip have also > been moved into their own file (dialplan_functions). > > Information that can be retrieved: > * rtp,type,[media_type] - Get RTP information, including media > source/destination addresses, whether or not the media is secure, etc. > * rtcp,statistic,[media_type] - Get RTCP statistic information > * endpoint - Get the name of the endpoint associated with this channel. Use > PJSIP_ENDPOINT to get more info. > * pjsip,type - Get signalling related information, including > source/destination addresses, URIs in the INVITE request, whether or not the > signalling is using a secure transport, etc. > > Note that after this patch is committed, we should go back through the > CHANNEL function documentation and move all of the channel technology > specific information into <info/> blocks, so that the documentation is > co-located with the channel drivers themselves. > > > Diffs > ----- > > /branches/12/res/res_pjsip_t38.c 403470 > /branches/12/include/asterisk/res_pjsip_session.h 403470 > /branches/12/funcs/func_channel.c 403470 > /branches/12/channels/pjsip/include/dialplan_functions.h PRE-CREATION > /branches/12/channels/pjsip/include/chan_pjsip.h PRE-CREATION > /branches/12/channels/pjsip/dialplan_functions.c PRE-CREATION > /branches/12/channels/chan_pjsip.c 403470 > /branches/12/channels/Makefile 403470 > > Diff: https://reviewboard.asterisk.org/r/3038/diff/ > > > Testing > ------- > > See https://reviewboard.asterisk.org/r/3037 > > > Thanks, > > Matt Jordan > >
-- _____________________________________________________________________ -- 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
