search for: directrtp

We're creating a SIP gateway for a client that will take one leg of a call in via SIP, and out the other side via H.323. To minimize load on the gateway, we would like to have the RTP stream bypass the gatewayy altogether (directrtp/reinvite). Is this possible with these to protocols? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100223/c067da2d/attachment.htm

...You go for splitting everything up, You could just simply drop in more machines. I think it would be more cost-effective to have 8 machines with 2 cores each. and that would additionally provide better I/O performance. Anyway, You can try throwing those calls and see how much can You get. As for directrtp=yes - i'm not sure what it does, but perhaps it's meant to be canreinvite=yes? Set it for each peer, and make sure You dial to peer, not to IP (as I recall - this didn't work globally) Regards, Atis On Wed, Apr 22, 2009 at 10:31 AM, Venefax <venefax at gmail.com> wrote: > Ye...

...o local ip through router as INVTE is meant for router ip and asterisk does not know where to send rtp stream after sending it to router. how can this issue be resolved? is there something to be done at router confiurations or sip.conf parameters. I have already played with nat/qualify/canreinvite/directrtp/externip etc parameters. regards, Nasir Javaid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100803/06e05a6e/attachment.htm

Hi All, I have a question about how a particular situation would work between two PBX systems: If I were to connect my Mitel box to my Asterisk box via a SIP trunk (same rack, same network), and then pass a call from the Mitel to Asterisk to perform some functions (lookups, maybe routing), and then pass the call back to the Mitel to be routed to it's endpoint, would Asterisk stay in that

...gt; and > > asterisk does not know where to send rtp stream after sending it to > router. > > > > how can this issue be resolved? is there something to be done at router > > confiurations or sip.conf parameters. I have already played with > > nat/qualify/canreinvite/directrtp/externip etc parameters. > > > > regards, > > > > Nasir Javaid > > > > > > ------------------------------ > > Message: 13 > Date: Tue, 03 Aug 2010 13:21:23 +0200 > From: Philipp von Klitzing <klitzing at pool.informatik.rwth-aachen.de> &gt...