Hi Olivier > Are those PBXs directly connected to each other through a SIP trunk ?
Yes, and the reason is definitely transmitted in the SIP header and also parsed by asterisk and displayed in debug output. 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. Kind regards Benoit Panizzon -- I m p r o W a r e A G - ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 Pratteln Fax +41 61 826 93 02 Schweiz Web http://www.imp.ch ______________________________________________________ -- _____________________________________________________________________ -- 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
