[Asterisk-Dev] Thoughts on a context for dialplan execution
upon a sip or iax register?
John Todd
jtodd at loligo.com
Sun Apr 24 12:07:09 MST 2005
At 9:21 AM -0700 on 4/24/05, Kevin P. Fleming wrote:
>
>It's an interesting thought, although obviously you wouldn't be able
>to run any applications that handle a media stream.
>
>What do you see being a potential use for this functionality?
I see this as a good idea, and I would add to this suggestion. I
would say that both registration and UNregistration events should
call that context, perhaps with different prefixes. Perhaps a
registration event causes "r1234" to be the extension called, and
unregistration causes "u1234" to be called, where "1234" is the
extension that is registering. Stripping off that first
character/digit should be fairly easy to make decisions.
I could see several things off the top of my head that would be
possible with this feature:
- when my laptop SIP client registers, all my calls go to my mobile
SIP client and no longer ring on my desk by a previously-set value
via a DBPut from the registration (this is a different functionality
than using SIP/1234&SIP/9876, since only one line rings at any time)
- my deskphone should always be registered. When my deskphone falls
out of registration, that is bad. When my phone comes back on-line,
I would like to immediately get a call from an automated system that
says "Your phone was unregistered for X minutes due to network
problems"
- Also in the previous example, when my deskphone falls out of
registration, I'd like to get a call from the system (via a Zap
channel) to my cellphone that my deskphone has gone away.
- I'd like to test ${SIPUSERAGENT} on registration, to see if I have
"registration fights" between two devices which are configured with
the same username/password. Instead of doing this on the INVITE, I'd
like to do it on the REGISTER.
NOTE: Some of these examples could be implemented with the
"regcontext=" feature and some acrobatic dialplan work. I find the
regcontext= feature to be useful, but not "real-time" in nature.
(http://www.voip-info.org/wiki-Asterisk+sip+regcontext)
JT
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