On Thu, 2006-07-06 at 23:22 -0300, Fabio wrote: > are you using SIP reinvite ?
Proably not as I'm using "t" in Dial()s for call transfer. > post a bit more information (sip.conf) [general] context=sip allowguest=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no domain=mydomain.com domain=1.2.3.4 allowexternalinvites=no language=it relaxdtmf=yes [authentication] [as5350] ; My PSTN gateway type=peer qualify=200 host=1.2.3.5 fromdomain=1.2.3.5 insecure=very [ser] ; My SIP proxy type=peer qualify=200 host=1.2.3.6 fromdomain=1.2.3.6 insecure=very [01]; Extension example callerid=My Name <01> nat=yes type=friend username=01 secret=mypass host=dynamic dtmfmode=rfc2833 context=uffici canreinvite=no callgroup=1 pickupgroup=1 qualify=no Thanks -- Luca Corti PGP Key ID 1F38C091 Adesso dico: "L'usignolo chiuso in gabbia smette di cantare." _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users