I would like to get call control of a telephone with PJSUA base and CSTA PBX. Thats mean, dont use pjsua as a softphone even use it for manage a hardware IP phone controlled by an IP-PBX, that support SIP uaCSTA. The objetive is make and hangup calls with the PC instead of the phone. Please even this mail is a bit large I have tried to explain detailed how I suppose I could obtain this operation with pjsua/pjsip, and I need some developers help. PJSIP doesnt support uaCSTA, but I suspect that I can make "call control" with some INFO request (I really only need dial and hangup in order to make "clic to call" funcionality). My problem is that first I have to initiate a call session with an INVITE with some specific header. In page 7 of the uaCSTA specification ( http://www.ecma-international.org/publications/files/ECMA-TR/TR-087.pdf ) says: " (...) The application creates an application session by establishing a SIP dialog with the user agent using a SIP INVITE method that includes a Content-Disposition header indicating ?signal? and "handling=required" to mandate support for the application/csta+xml MIME type. An ECMA-323 Request System Status service request is included in the SIP INVITE body with the Content-Type application/csta+xml. (...)" I have captured (from wireshark) an INVITE packet requesting this control from one working (and closed source) softphone, as follow: Session Initiation Protocol Request-Line: INVITE sip:pbx at company.com <sip%3Apbx at company.com> SIP/2.0 Method: INVITE [Resent Packet: False] Message Header v: SIP/2.0/UDP 10.151.86.95:5060;branch=z9hG4bK49d4d1e9-9 Transport: UDP Sent-by Address: 10.151.86.95 Sent-by port: 5060 Branch: z9hG4bK49d4d1e9-9 i: 002b-0bb0-066c-ffffffff at 10.151.86.95 Max-Forwards: 40 t: <sip:pbx at company.com <sip%3Apbx at company.com>> SIP to address: sip:pbx at company.com <sip%3Apbx at company.com> f: <sip:segalion at company.com <sip%3Asegalion at company.com> >;tag=2b-bb0-66c-ffffffff SIP from address: sip:segalion at company.com<sip%3Asegalion at company.com> SIP tag: 2b-bb0-66c-ffffffff CSeq: 1 INVITE Sequence Number: 1 Method: INVITE Accept: application/csta+xml Expires: 180 m: <sip:segalion at 10.10.86.95:5060> Contact Binding: <sip:segalion at 10.10.86.95:5060> URI: <sip:segalion at 10.10.86.95:5060> SIP contact address: sip:segalion at 10.10.86.95:5060 User-Agent: Unknown Allow: ACK, MESSAGE, NOTIFY, INVITE, BYE, CANCEL, REFER, OPTIONS, INFO, PUBLISH, UPDATE Accept-Language: es-es k: replaces x-nt-GUID: 0034a185170bdc304b150a9ad08ab288691bb0 x-nt-location: 35395728 P-Preferred-Identity: <sip:segalion at company.com<sip%3Asegalion at company.com> > c: application/csta+xml Content-Disposition: signal; handling=required l: 205 Message Body <?xml version="1.0" encoding="UTF-8"?><RequestSystemStatus xmlns=" http://www.ecma-international.org/standards/ecma-323/csta/ed3"; xmlns:xsi=" http://www.w3.org/2001/XMLSchema-instance";></RequestSystemStatus> My problem is to construct this headers with pjsip. I was reading Developers Guide, but I need some orientation about making a call INVITE with this headers (basically "content_type: application/csta+xml" and "Content-Disposition: signal; handling=required") and no SDP media negotiation, but I want to mantain the automatic control state on pjsua core (acks, timeouts, etc.). I was thinking in make a new "pjsua_call_make_call" function modification without sound, media and sdp parts, and a specific pjsua_process_msg_data. Could anybody help me with all this (orientation of how construct this header with minimum modification of pjsua code). I suppose when I have the "200 OK" I can costruct my own CSTAs INFO requests to make call control. Please, help me. I need help from developers, and I dont know if this mailing list is the best site. Thanks in advance... Segalion -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090623/d4a70c7e/attachment.html>
