We already have a stub mod_pjsip.c with promise of a GPL exception for non-commercial use and licensing options available for commercial use. I also have a mod_sipx.cpp lying around somewhere and of corse good ol' mod_exosip. If you want to play with any of them let me know. FreeSWITCH is designed to have mutiple endpoints so as long as they do not try to use the same IP/Port they can co-mingle. Be aware that we have tried many solutions and sofia to date has been able to provide the most reliability with pjsip comming in as a 2nd place hopeful but seeming to fail on some race conditions and not guaranteed to work right on 64bit. Perhaps pick some less-developed area of the project to enhance =D Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: "ed@edpimentel.com" <ed@edpimentel.com> To: freeswitch-users@lists.freeswitch.org Sent: Thursday, March 8, 2007 6:48:40 AM Subject: Re: [Freeswitch-users] Click2Call question Is it possible to use plug in another SIP-STACK ie http://www.pjsip.org/pjsip/docs/html/index.htm Or you can also dovetail FS with http://www.jivesoftware.com/products/wildfire/featuretour.jsp And use Smack API and XPP http://www.igniterealtime.org/ http://www.extreme.indiana.edu/xgws/xsoap/xpp/ The Holy Grail would be when we can plug in Jay's AdHEARsion engine http://adhearsion.com/ to FS To that point, let me know who would want to sponsor with me a port of AdHEARsion to FS with me. Best, -Edpimentl

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Hi,

...

You have to remember that FreeSWITCH is not a sip stack... Sofia-SIP is. We will need to raise this issue with them.

hm you are right ... :)

...

Why is it sending an invite without an SDP?

Well dunno. I guess it's because a webapplication starts the INVITES to both caller and callee and for the first one to Caller the application has no idea about the SDP data for caller. the SDP is send in a 200 OK message from caller to freeswitch and from there back to web application. After that the web app invites the calle with sdp from caller ...

regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFF7tW34tZeNddg3dwRAmaXAJ9RWyGBVw/O0hB0ai8qi/zT82yDRQCfYleI isydW+Wgdl1hed44SZOqAy4= =ZUel -----END PGP SIGNATURE-----

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