[asterisk-users] RTP traffic through Asterisk??

2009-11-13 Thread Ignacio
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 res

Re: [asterisk-users] RTP traffic through Asterisk??

2009-11-13 Thread John A. Sullivan III
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 bot

Re: [asterisk-users] RTP traffic through Asterisk??

2009-11-14 Thread Ignacio
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 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 R

Re: [asterisk-users] RTP traffic through Asterisk??

2009-11-15 Thread Leonja Cerebro
see the DTMF method on both phones. 2009/11/14 Ignacio > 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 > wrote: > > On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote: > >> I have just e

Re: [asterisk-users] RTP traffic through Asterisk??

2009-11-17 Thread Ignacio
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 p

Re: [asterisk-users] RTP traffic through Asterisk??

2009-11-17 Thread Aggio Alberto
on 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 trans

Re: [asterisk-users] RTP traffic through Asterisk??

2009-11-24 Thread John A. Sullivan III
Can you move the transfer functionality to the end device rather than through Asterisk? That's what we do - John On Tue, 2009-11-17 at 14:07 +0100, Ignacio wrote: > 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