Kevin, I had no clue! Thanks for the note! I'll be checking out the 1.8 SVN branch here shortly then for testing!
Thanks again! Louis -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Kevin P. Fleming Sent: Monday, March 07, 2011 9:34 AM To: [email protected] Subject: Re: [asterisk-users] Asterisk <-> Lync / Call Center Transfer / Refer On 03/04/2011 12:35 PM, Louis Carreiro wrote: > Ha! Thanks Vip! > > Sorry about not including my version numbers too. On my production box I'm > using 1.8.3 (that's the debug from the original email). On my demo box I just > build I'm using 1.8 SVN-trunk-r309404 and that's what generated these logs. > I'm not sure if this is a chan_sip.c problem or if this is a dial plan > problem. If your version string is 'SVN-trunk-r309404', you are not using 1.8, you are using 'trunk'. If you want to follow the 1.8 Subversion branch, you need to checkout that branch, not trunk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: [email protected] | SIP: [email protected] | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
