In article <[EMAIL PROTECTED] aachen.de>, [EMAIL PROTECTED] says... > It is not only re-invite that determines what happens to your media path, > there are also Dial() arguments like t,T,w,W (and possibly some more) > that can force it go through Asterisk. The same applies to codec > settings, i.e. if you need Asterisk in between to transcode e.g. from > g729 to alaw then obviously the rtp stream has to go thru Asterisk.
Thank you for explaining this to me. > Next to that: Try to switch both your phone and your Asterisk config to > dtmfmode=info (SIP INFO) and see if automon recording will work that way > even if you have canreinvite=yes - it could work since in this case DTMF > is transmitted as SIP message; I have to admit that I am not 100% sure if > with canreinvite=yes Asterisk will also be completely cut off from the > SIP signalling stream, but I think it'll still be in the loop - haven't > tried it myself. I have found out that the problem is with Cisco phones (7905 and 7940). With ZAP (analog) phones and with Grandstream it works fine. I'm opening new thread for this one => Cisco dtmf > For your transfer question: You'll have to use t or T in Dial in order to > permit transfer, which in turn means your rtp traffic will be forced thru > Asterisk no matter what your canreinvite= settings looks like. > You might want to look at if and how your SIP phone supports native > transfer by itself. My phones support nativ transfer but I would like to implement this feature. -- Tomislav Parcina [EMAIL PROTECTED] _______________________________________________ --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
