On Wed, Nov 14, 2018, at 12:01 PM, Hans-Peter Jansen wrote: > [Sorry for resending, but I have to in order to conform to the CLA] > > @Joshua: now that I signed the CLA, may I kindly ask you to look at the code > now? Ideas, where/how to get/synthesize call-id, local-tag and remote-tag > attributes and the caller info in PJSIP land are appreciated. > > @everyone: this is an attempt to implement Dialog-Info+XML for PJSIP to get > on > par with chan_sip. It will allow subscribed SNOM phones to display the caller > > to some monitored extensions and to pick that call up by pressing the > corresponding function key without distributing the call in call groups. In > Asterisk 1.2 times, this was implemented by the bristuff patches. chan_sip of > current Asterisk versions support this feature out of the box. > > On Montag, 29. Oktober 2018 13:56:17 Joshua C. Colp wrote: > > On Mon, Oct 29, 2018, at 1:47 PM, Hans-Peter Jansen wrote: > > > Dear Asterisk developers, > > > > > > in an attempt to add the missing pieces in > > > res/res_pjsip_dialog_info_body_generator.c to provide a similar > > > Dialog-Info+XML implementation, as what chan_sip.so provides already, > > > I invested the better part of today, but things seem to be much more > > > complicated in PJSIP land (at least for somebody, who started to look > > > at this code today). > > > > > > This is the only missing functionality, that keeps me from transitioning > > > to PJSIP, and, if I read the various related complains correctly, a lot of > > > other Asterisk users as well. > > > > > > What I found out so far: > > > > > > PJSIP version: > > > > > > <?xml version="1.0" encoding="UTF-8"?> > > > <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="3" > > > state="full" entity="sip:62@192.168.23.2:15060"> > > > > > > <dialog id="62" direction="recipient"> > > > > > > <state>early</state> > > > > > > </dialog> > > > > > > </dialog-info> > > > > The information does not currently exist in PJSIP, 'nor does it get passed > > in. The chan_sip module has special logic (find_ringing_channel) local to > > it to gather the information it thinks is correct which is then placed into > > the message. The same kind of thing would need to be done in PJSIP. > > First of all, thanks for your instant response, Joshua. > > Here's, where I got with some hackery today (attached): > > <?xml version="1.0" encoding="UTF-8"?> > <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" > state="full" entity="sip:62@192.168.23.2:15060"> > <dialog id="62" direction="recipient"> > <remote> > <identity display>sip:000413414123@192.168.23.2</identity> > <target uri="sip:000413414123@192.168.23.2" /> > </remote> > <local> > <identity display="hp Office 2">sip:62@192.168.23.2:15060</identity> > <target uri="sip:62@192.168.23.2:15060" /> > </local> > <state>early</state> > </dialog> > </dialog-info> > > Remote is still wrong, it's a local extension, and I also have no idea ATM, > where to fetch call-id, local-tag and remote-tag attributes. It also makes > asterisk not to exit gracefully anymore after hitting ^C. > > May I kindly ask you to take a look at it?
The chan_sip module appears to use the information of the underlying subscription for the call-id and tags. This can be retrieved using the ast_sip_subscription_get_dialog function to get the dialog, and then accessed on it[1]. Specifically call_id, local, and remote. Can you clarify what you mean by "Remote is still wrong". As well on a cursory glance I don't see anything that would explain the ^C problem, but it's possible there is a memory issue within it that I don't see immediately. [1] https://www.pjsip.org/pjsip/docs/html/structpjsip__dialog.htm -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
