George Joseph
2021-Aug-20 14:14 UTC
[asterisk-users] Between a dumb client and a capable server...
On Wed, Aug 18, 2021 at 3:33 AM Antony Stone < Antony.Stone at asterisk.open.source.it> wrote:> Hi. > > I wonder if anyone has some helpful advice or suggestions for me? > >snip I had thought that Kamailio might be what I was looking for, but I've asked> on > their mailing list and people are telling me that it isn't, and that I > need > something like Asterisk to do this. I'm trying to get some specifics from > them > about *how* I would get Asterisk to do this (because I personally can't > see > how Asterisk could sit between a SIP client and a SIP server, and generate > commands to manipulate the RTP stream and send them to the server, which > is > what the Kamailio people are saying I should do), but I thought it was > worth > asking here just in case what they're telling me is in fact quite easy > when > you only know enough about Asterisk. > > So, if someone here thinks this is possible using Asterisk, please could > you > point me at some documentation showing what commands I would use or the > basics > of how I should go about it? > > If anyone thinks there is another, perhaps better, way of achieving this, > then > I'm quite open to alternative solutions (as I say, I was initially > thinking > that Kamailio might be the way forward), so anything that shows me *how* > such > a thing might be achieved, with any tool at all, would be very welcome. > > 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). It also sounds like the 'SIP server" isn't Asterisk and you can't change that either right? You could actually use a tiny Asterisk instance to do this. The dumb client would call Asterisk and Asterisk would simply send the call to your existing SIP server. 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. The big question is... what triggers the action to take? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20210820/4874c29a/attachment.html>
Antony Stone
2021-Aug-20 14:33 UTC
[asterisk-users] Between a dumb client and a capable server...
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')? 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.