On 15-10-19 09:12 AM, Andrew Colin wrote:
Do you know if this can be achieved with the standard sip stack in asterisk?
If you are referring to chan_sip I don't believe so but it is possible there is some obscure option or method to do it that I am aware of.
-- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
