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

[Louis Carreiro]

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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Monday, March 07, 2011 9:34 AM
To: asterisk-users at lists.digium.com
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
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org

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