1 sep 2009 kl. 05.18 skrev John A. Sullivan III: > On Thu, 2009-08-27 at 14:23 -0400, John A. Sullivan III wrote: >> Hello, all. In our multi-tenant environment, we would like to be >> able >> to use the reinvite media redirection within Asterisk for calls >> within a >> tenant but not between tenants. We would like inter-tenant calls >> to be >> fully proxied by the Asterisk server. I think the answer is, "we >> can't," but I thought I'd ask anyway. >> >> I'd dearly like to remove the substantial traffic associated with >> intra-tenant traffic from the Asterisk server and reduce the >> intra-tenant latency by doing so. However, I am very, very >> hesitant to >> allow our VPN connections to tenants to function as a router between >> tenants allowing one tenant to directly access phones on another >> tenant >> (that's not as wild as it sounds because of our use of the ISCS >> project >> - iscs.sourceforge.net). >> >> Since the tenants are all connecting via VPN, we are using RFC1918 >> addresses and no NAT is involved thus the canreinvite=nonat option >> does >> not help us. If we set canreinvite=nonat, that will allow for >> intra-tenant direct media but, if one tenant tries to call another >> via >> SIP, it will redirect the media at the Asterisk level but the packets >> will be dropped at the firewall / router level (or sooner as there >> may >> be no route to the destination) and the call will connect but with no >> sound. >> >> Any guidance would be greatly appreciated. Thanks - John > > As mentioned in another post, we were able to solve this by setting > a w > dial option to all inbound SIP calls from the Internet. Thus, all > internal calls could reinvite but external calls could not. > > However, just when we thought this was working splendidly well, we > turned up another roadblock - transfers. We find that once we > transfer > a call using our Snom phones, the call between the new call partners > does not seem bound by the "w" constraint, Asterisk tries to reinvite > the call, and the audio breaks because the firewall cannot associate > the > new RTP stream with a SIP session. > > How have others gotten around the problem of transfers causing > reinvites > on calls which should not allow reinvites? Thanks - John
I think this is an issue that needs some code to solve it, so you can set a variable in the dialplan that prevents remote RTP bridges (reinvited media). Contact me off list if you're interested in sponsoring such development. /O _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
