One thing I haven't seen get much airtime on the digium lists is sip
URL-based peering. I imagine many of us have far more asterisk
extensions than PSTN numbers. It would be really nice to be able to
do something like call 7123@joes-asterisk from 2123@my-asterisk. It
looks like all or most of the pieces are in place, but I don't see
folks discussing it much. Is no-one else interested in this?
One group that seems to have an ever growing list of sip servers that
accept direct incoming sip calls is sipbroker. Using their service
doesn't really buy the average asterisk admin much, but they do have a
nice list of sip servers and they do assign a unique prefix code to
each server which might be useful to snarf into an asterisk database.
http://www.sipbroker.com/sipbroker/action/providerWhitePages
extensions.conf:
;; send everything else with a ** prefix to Sipbroker
;; strip one of the stars since they only want one in total.
;; http://www.sipbroker.com/sipbroker/action/providerWhitePages
exten => _**XXX.,1,Macro(dial-outgoing,SIP/${EXTEN:1}@sipbroker-out)
sip.conf:
;;; outgoing to Sipbroker
[sipbroker-out]
type=peer
host=sipbroker.com
Is sipbroker just a well-kept secret from the asterisk crowd, or is
everyone else using asterisk for phone spamming from call centers and
the last thing they want is folks to be able to call them back and
give them an earful over disturbing their dinner?
-wolfgang
--
Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/