On Fri, Aug 20, 2021 at 8:33 AM Antony Stone < antony.st...@asterisk.open.source.it> wrote:
> On Friday 20 August 2021 at 16:14:44, George Joseph wrote: > > > On Wed, Aug 18, 2021 at 3:33 AM Antony Stone wrote: > > > Hi. > > > > > > Just to summarise: I have a SIP client talking to a SIP server, and I > > > need something which can send commands to that server to put calls, > > > which were created by the existing client, on hold (that's the simplest > > > scenario). I do not want to build a SIP server / PBX myself which can > > > itself perform call hold & transfer etc (I know how to do that with > > > Asterisk) - I need those functions to be performed by the existing > server. > > > > Sounds like you're looking for something to do 3rd Party Call Control > > (3PCC). > > Okay, that sounds like useful terminology. > > > It also sounds like the 'SIP server" isn't Asterisk and you can't change > > that either right? > > It *might* be Asterisk, but if it is, I have no access to it other than > the > SIP credentials a standard telephone would use to register to it. Then > again, > I might not even *know* what it is - it's just a SIP-based PBX... > > > You could actually use a tiny Asterisk instance to do this. > > Hm, I'm very dubious about that, based on what I've seen in docs so far... > > > The dumb client would call Asterisk and Asterisk would simply send the > call > > to your existing SIP server. > > Okay, so far, so good, I can get Asterisk to do that. > > > You could then use AMI or ARI to watch for the call events and tell > > Asterisk to transfer to some other extension on your SIP server or > whatever. > > So, let's just take the simplest example - how can I get Asterisk to tell > the > other server to put a call on hold and play that other server's hold music > to > the remote party? > > > The big question is... what triggers the action to take? > > That's easy, I have a web interface which is on the same machine as the > dumb > SIP softphone, and that can talk to this "tiny Asterisk server" you > speculate > about, for example by sending in AMI Originate commands to it, which can > trigger dial plan actions, which can do anything Asterisk is capable of. > > My doubts are whether Asterisk as a SIP *client* is capable of this. > > 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. > > > Thanks, > > > Antony. > > -- > "The future is already here. It's just not evenly distributed yet." > > - William Gibson > > Please reply to the > list; > please *don't* CC > me. > > -- > _____________________________________________________________________ > -- 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
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