As far as I could try some solutions, the only one that works as you like involved use of Transfer() application, defining as 'tecnology' something like that:
SIP/<exten>@<ip_address> Where <ip_address> is the address of the peer you want to transfer the call to. By the way, I found a scenario where this trick still keeps not working: if the transferor (i.e. the caller) is a registered SIP user, I saw that the transfer is done, but Asterisk is still in the path. Vice versa, if the caller is NOT a registered user, the transfer will exclude asterisk from the path either if the transferree (i.e. third party called) is registered to Asterisk or not. HTH Alberto. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ignacio Sent: martedì 17 novembre 2009 14.07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic through Asterisk?? Thank you very much to both of you. My problem was that I used transfer in the dialplan. I have read that If I have Tt, wW, or hH, then asterisk will always stay in the path. So I have to redefine what I want to do know. Allowing transfers is an useful feature, but I wanted all rtp traffic went p2p. Is there any intermediate solution? Thanks. Regards Ignacio On Mon, Nov 16, 2009 at 7:52 AM, Leonja Cerebro <lio...@gmail.com> wrote: > see the DTMF method on both phones. > > 2009/11/14 Ignacio <sanfermi...@gmail.com> >> >> Ok, thank you very much. I will try to find any information in >> asterisk documentation about RTP. >> >> On Fri, Nov 13, 2009 at 3:03 PM, John A. Sullivan III >> <jsulli...@opensourcedevel.com> wrote: >> > On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote: >> >> I have just established a call between 2 sip phones and I have noticed >> >> that all RTP traffic goes through Asterisk Server. >> >> >> >> I was expecting RTP traffic went to one phone to another phone >> >> directly. >> >> >> >> I set canreinvite=yes in sip.conf in both sip peers. >> >> >> >> I also tested it with 2 mgcp phones and same result, all rtp traffic >> >> goes through Asterisk. >> >> >> >> Is there any way to force traffic to go from one phone to another? >> > <snip> >> > I don't recall where it is off-hand but, somewhere in the Asterisk >> > documentation, there is an explanation of how Asterisk makes a decision >> > about reinvites. You may want to look at that to see if your >> > environment satisfies all the requirements and how it can be adapted if >> > it does not - John >> > -- >> > John A. Sullivan III >> > Open Source Development Corporation >> > +1 207-985-7880 >> > jsulli...@opensourcedevel.com >> > >> > http://www.spiritualoutreach.com >> > Making Christianity intelligible to secular society >> > >> > >> > _______________________________________________ >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > We never did too much talking anyway > So don't think twice, it's all right > ---------------------------------------------------------- > There are more things in heaven and earth, Horatio, > Than are dreamt of in your philosophy. > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users