On Wednesday 18 August 2021 at 14:19:56, Andreas Wehrmann wrote: > If I understand correctly, you want to be able to (un-)hold and transfer > calls. Correctly understood :) > If you want to hold/transfer "your" calls, you can do so already (like > any other SIP phone). > > For holding/unholding calls, see the functions > pjsua_call_set_hold()/pjsua_call_set_hold2() here: > https://www.pjsip.org/docs/latest-2/pjsip/docs/html/group__PJSUA__LIB__CALL > .htm#gabc6cf34d2b241224230701025eef489a > > For transferring calls, you can initiate a transfer via REFER using i.e. > pjsua_call_xfer(): > https://www.pjsip.org/docs/latest-2/pjsip/docs/html/group__PJSUA__LIB__CALL > .htm#ga97637997a55ef0246aae5b3d171532f2 Okay, that sounds excellent - thank you :) Where should I start with a guide to how to get PJSIP installed into my network path so that it sees the call setups between the SIP client and the server, and I can start using these functions to manage the calls? Sorry for such a basic question, but I have never used PJSIP before, so a pointer in the right direction to get me started would be most helpful. If it matters, my SIP client is currently running under Debian Linux, and I'd be happy to install PJSIP on the same machine, or another one in the network, whichever is best. > If you want to manage all kinds of calls on your PBX, then this is > outside the scope of the SIP protocol and you need to check > with the PBX developers. I don't want to do anything which a standard SIP telephone can't do, therefore I am only looking to do things within the scope of SIP. > In my experience, they usually offer some kind of "management" > interface/protocol which you can use to control the PBX. For an Asterisk > PBX, this would be the AMI and ARI interfaces. Well, yes, and in some cases the PBX I am talking to will be Asterisk, but after asking about this topic on the Asterisk mailing list, I got the simple confirmation from Joshua Colp, one of the Asterisk developers, that call hold & resume simply isn't possible using AMI. I find it strange, since AMI will happily *notify* you of calls being put on hold or resumed, but there is no way to use it to *put* a call on hold. However, as I say, I'm not trying to do anything outside the scope of SIP, and I would really prefer not to have to implement a different way of doing it for every SIP-based PBX I end up having to register an extension to. Many vendors would not even tell me what access credentials I could use to connect in the first place, whereas I always know the SIP credentials for the extension assigned to my client application. Thanks, Antony. -- Perfection in design is achieved not when there is nothing left to add, but rather when there is nothing left to take away. - Antoine de Saint-Exupery Please reply to the list; please *don't* CC me. _______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list -- pjsip@xxxxxxxxxxxxxxx To unsubscribe send an email to pjsip-leave@xxxxxxxxxxxxxxx
