Summary:	ASTERISK-21257: Implement inbound/outbound Caller ID handling
Reporter:	Matt Jordan (mjordan)	Labels:	Asterisk12 NewSIP
Date Opened:	2013-03-15 08:30:17	Date Closed:	2013-04-15 17:23:39
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_pjsip
Versions:	12	Frequency of Occurrence
Related Issues:	must be merged before resolving ASTERISK-21258 Implement mid-call connected line support for chan_gulp
Environment:		Attachments:
Description:	Hey, we have basic calls! Now we should know *who* is calling. Ideally, {{chan_gulp}} would be responsible for very little of this, and the vast majority of the application level logic that decides how Caller ID is represented to the rest of Asterisk for a particular call would be placed into its own {{res_*}} module. This should let parties that want to modify the application logic to either: * Easily modify it outside of the channel driver/SIP logic * Roll their own module that provides caller ID determination * Allow other SIP modules to obtain caller ID information for the particular call. (It may be better to get this off the channel object, but this should allow us to support SNOM's caller ID in SIP NOTIFY features) The following legacy functions should be supported in some fashion: * shrinkcallerid - remove '(', ' ', ')', and non-trailing '.' and '-' not in '[]' * trustrpid/sendrpid * callerid (what we send) (Note: I'm putting connectedline as its own thing)
Comments: