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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-dev-bounces@lists.digium.com [mailto:asterisk-dev-bounces@lists.digium.com] <b>On Behalf Of </b>George Joseph<br><b>>Sent:</b> Wednesday, October 01, 2014 1:16 PM<br><b>>To:</b> asterisk-dev@lists.digium.com<br><b>>Subject:</b> [asterisk-dev] Opinions needed: Should PJSIP support enhancements still be allowed in 12->13?<o:p></o:p></span></p></div><p class=MsoNormal><span style='color:#1F497D'>></span><o:p> </o:p></p><div><p class=MsoNormal><span style='color:#1F497D'>></span>Even though I've been using pjsip on my dev and test machines for a year, it's only in the last few weeks I've tried to implement pjsip in a prod environment and that's brought up some <span style='color:#1F497D'>></span>issues. Unfortunately now that we're close to a 13 release there's some trepidation about addressing these issues in 12->13 as opposed to just trunk...<o:p></o:p></p><div><p class=MsoNormal><span style='color:#1F497D'>></span><o:p> </o:p></p></div><div><p class=MsoNormal><span style='color:#1F497D'>></span>res_phoneprov: Today's phoneprov infrastructure is strictly chan_sip using users.conf and sip.conf for all user related configuration. There's no pjsip support there at all. I have 2 patches posted <span style='color:#1F497D'>></span>to RB ([1], [2]) that make res_phoneprov pluggable and provide a pjsip provider for phoneprov. All existing functionality remains unchanged. You just get the same capabilities for pjsip that <span style='color:#1F497D'>></span>you had for sip.<o:p></o:p></p></div><div><p class=MsoNormal><span style='color:#1F497D'>></span><o:p> </o:p></p></div><div><p class=MsoNormal><span style='color:#1F497D'>></span>manager/config: From a remote configuration management perspective The AMI commands GetConfig, GetConfigJSON and UpdateConfig allow you to manipulate Asterisk config files, BUT <span style='color:#1F497D'>></span>they don't work well on configurations that can have multiple sections with the same name. This was rarely a problem before pjsip but now you can have an endpoint, an aor, an auth, an identify <span style='color:#1F497D'>></span>and an registration all named 'myitsp'. This makes it impossible to manipulate them via AMI. I have a patch to enhance manager.config with that capability as well as the cabability to <span style='color:#1F497D'>></span>manipulate config templates via AMI. [3]<o:p></o:p></p></div><div><p class=MsoNormal><span style='color:#1F497D'>></span><o:p> </o:p></p></div><div><p class=MsoNormal><span style='color:#1F497D'>></span>Finally, While thinking of alternatives for the config file dilemma, Josh, Brad and I tossed around the idea of a 'super' pjsip object or objects that could represent a 'trunk' or a 'user' thereby <span style='color:#1F497D'>></span>eliminating having to specify separate endpoint, aor, auth, identify and registration objects for common scenarios. [4] I just started writing code for this today.<o:p></o:p></p></div><div><p class=MsoNormal><span style='color:#1F497D'>></span><o:p> </o:p></p></div><div><p class=MsoNormal><span style='color:#1F497D'>></span>So now the big question is... Can these items go into 12/13 or should they go only in ttrunk/14. I do need these patches but I can always apply them to my own distro. My opinion though is <span style='color:#1F497D'>></span>that they should go into 12/13 to help speed the adoption of pjsip. No one who uses phoneprov or AMI to manipulate config files will able migrate otherwise. Having said that, I realize that 13 <span style='color:#1F497D'>></span>GA is almost upon us and having defined cutoffs is a very good thing.<o:p></o:p></p></div><div><p class=MsoNormal><span style='color:#1F497D'>></span><o:p> </o:p></p></div><div><p class=MsoNormal><span style='color:#1F497D'>></span>Thought's? Opinions? <span style='color:#1F497D'><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>If PJSIP in Asterisk 13 is *<b>supposed</b>* to have feature parity with chan_sip, then I’d call the above items bugs rather than enhancements.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I strongly support the policy of “no new features” after a branch has been released, but sometimes the changes are so important they should be made anyway. AELSub is a great example of the latter.<o:p></o:p></span></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div></div></div></body></html>