Summary:	ASTERISK-20860: Create chan_gulp - Phase 1
Reporter:	Matt Jordan (mjordan)	Labels:	Asterisk12 NewSIP
Date Opened:	2013-01-03 13:33:34.000-0600	Date Closed:	2013-01-24 09:11:00.000-0600
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Channels/chan_pjsip
Versions:	12	Frequency of Occurrence
Related Issues:	cannot be resolved before merging of ASTERISK-20691 Create a generic thread pool for Asterisk cannot be resolved before merging of ASTERISK-20692 Implement a Data Access Layer for the new SIP channel driver cannot be resolved before merging of ASTERISK-20810 Create res_sip
Environment:		Attachments:
Description:	The first phase in creating {{chan_gulp}} is to get a channel driver in place that can make two party calls. The following should be met: * The channel driver should make use of the architecture outlined on the wiki page: (https://wiki.asterisk.org/wiki/display/AST/New+SIP+Channel+Driver+Architecture) * The channel driver should make use of {{res_sip}} and {{sorcery}} for it's object creation/persistence * The SIP endpoints do not have to register or be authenticated. Configuration of the endpoints can come from an in-memory provider. A team branch should be created for this.
Comments: