[asterisk-users] Asterisk <-> Lync / Call Center Transfer / Refer

Kevin P. Fleming kpfleming at digium.com
Mon Mar 7 08:34:17 CST 2011


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: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org



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