On 11-05-20 10:39 AM, Benoit Panizzon wrote: > After reading a bit more in chan_sip.c (I'm not a coder) I fear the result is > just put in a temporary variable __SIPDIVERSIONREASON but not in a variable > useable in the dialplan.
You could double check by using DumpChan() to see what channel variables are available for you throughout the dialplan flow. Also check the CHANNEL() and SIP*() functions to see if there is anything there that may be of use. Leif. -- _____________________________________________________________________ -- 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
