This is wordy but you only need to read the Problem section to answer my question, and if you read on, a simple ‘you are on the right track’ might suffice.
Sadly, becomming “pretty good” at administrating asterisk hasnt helped me undserstand the sip protocol. so, Im reading up on SER and SIP, but I needed some help understanding how the asterisk side is going to work in order to “wrap my mind around” a solution to my problem
the problem: i administrate asterisk servers at several locations for a particular business, each with its own NAT and sip clients. obviously SIMPLE is not supported and i must log every client into every server to hack working presence.
Possible solution: i need some way to implement SER (preferably only once at yet another NAT’d location) to act as a proxy. I feel this will work because since SER is a SIP Router, it should be able to “do the right thing” with an SIMPLE and Presence while keeping a voice session’s media stream in the control of the local * server.
each client is already logged into its respective * server. i hope to also simply register each client to SER for intranetwork presence and IM.
there is currently a prefixed based method that members of one * server can use to call the remote * server’s users. i basically envision using SER to redirect voice INVITES from the calling party’s to their local * server, in order to keep the media stream in the right place.
The only time this is really going to be needed is when my users naturally “hit the button” that corresponds to the pressence lamp or the contact they’ve been messaging. if i do nothing to intervene however, i may be able to get it working with the NAT, but I’ve lost the media stream, right?
i’d just like a general non-technical description of how asterisk will interface with SER, nothing specific, which i can keep in mind as i contiune to research. if you think you have a good solution please let me know what it is. Iif you’d like, read below to hear me continue to babble stuff that is probably wrong, but describes how i think its gonna work as of now.
i’m sure someone has come accross this problem and implemnted a good solution, but here is how i’m beginning to think it will work.
So i figure I will have to register => all of my asterisk servers with SER. each of these servers have their respective clients (snom phones, eyebeam softphones) which support presence, SIMPLE messaging, multiple identities, etc. each server has an globally resolvable URL. we can keep the example to two servers (ny_ast_server.com and cali_ast_server.com).
so jane in new york has an snom phone. its now logged into jane@ny_ast_server.com, and jon is in california where his snom is logged into jon@cali_ast_server.com.
jon’s internal extension is 100 and jane’s is 101. if jon wants to call jane he can dial 6101 and she can call him by dialing 4101.
so now i register a second ‘identity’ in both their snoms to jon@tex_ser_proxy.com and jane@tex_ser_proxy.com, and i can easily program a lamp on the snom to indicate presence on SER, but as I’ve stated above, i’ve lost control of the media as soon as my user presses that button to make the call.
instead, i hope to set up SER so that when jon presses the button the SNOM sends an INVITE to SER, which needs to send some 300 REDIRECT sip resonse and give the correct extension. what makes the most sense to me then, is that logic in ser.cfg will be able to point jon to 6101@ny_ast_server.com, thereby doing the right thing with the media. im guessing all i have to do is have calls from my SER user in the right context. Will that ‘just work’ with Sip and Asterisk?
if you’ve made it this far you are probably a better man than I thanks in advance for the anticipated responses.