On Fri, Aug 20, 2021 at 2:33 PM Eric Wieling <ewiel...@nyigc.com> wrote:
> > > On 8/20/21 4:24 PM, Antony Stone wrote: > > On Friday 20 August 2021 at 19:06:09, George Joseph wrote: > > > >> On Fri, Aug 20, 2021 at 8:33 AM Antony Stone wrote: > >>> > >>> So, if I have Asterisk registered as a SIP client to some remote > server, > >>> how can I get Asterisk to tell that remote server to put the call on > hold > >>> (which a standard SIP telephone would normally do by sending a ReINVITE > >>> with the SDP parameter 'sendonly')? > >> > >> On the outgoing pjsip endpoint, set "moh_passthrough = yes". If you > then > >> put incoming call on hold, a reinvite with sendonly will be sent to the > >> upstream server. > > > > So... how do I put the incoming call on hold, when the dumb client I'm > > starting from cannot do that bit? > > > > I already know (from this list) that Asterisk as a SIP client cannot do > ore > > than (a) place a call, (b) answer a call, and (c) hang up a call. > > > > So, I'm still intrigued as to how you think this might be possible. > > > > If it *is* possible, I'd be really interested, but all my researches so > far > > suggest that Asterisk, acting in the middle like this, just cannot add > the > > necessary "put call on hold" which the original client cannot do. > > > > With Asterisk, keep Asterisk in the media path with direct_media=yes and > use DTMF to hold, transfer, and other features using features.conf. > Asterisk has to stay in the media path when NAT is involved anyway. > You need to set direct_media=no to keep Asterisk in the media path. > > I doubt anything except Asterisk or other B2BUA software can do what you > want. > > -- > http://help.nyigc.net/ > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
