On 8/28/07, Stefan van der Eijk <[EMAIL PROTECTED]> wrote: > > > On 7/9/07, Noah Miller <[EMAIL PROTECTED] > wrote: > > Hi Stefan - > > > > > What I want to accomplish: > > > - calls within the LAN are re-invited (RTP goes from endpoint to > endpoint) > > > - asterisk detects when a call is going beyond the local LAN (over the > NAT), > > > and then stays in the middle. > > > > > > I'm wondering if this is hard to do and how I'm supposed to configure > this. > > > > I don't really know how hard it would be to do what you describe, but > > if you're interested in getting the results you want with a minimum of > > effort, just keep asterisk in the media path all the time. Set > > canreinvite=no, and your calls should work consistently whether they > > stay inside the NAT or go outside. > > This is what I ended up doing. Until I ran into issues again with outgoing > calls. Current setup = asterisk 1.4.11, installed on a host connected to the > internet (internet route able IP-address) and my internal network > (192.168.254.254). SIP phones are on the internal network, STUN and such > hasn't been configured. > > SIP.conf: > externhost = <external hostname --> ddns.org> > canreinvite = no > localnet = 192.168.254.0/24 > ; nat = option is not set > > Outgoing call to our sip provider ends up being setup like this: > > outbound RTP stream: > SIP phone (192.168.254.104) --> asterisk internal (192.168.254.254) > asterisk external (internet IP) --> asterisk external (internet IP) (!!!) > > inbound RTP stream: > SIP provider (internet IP) --> asterisk external (internet IP) > asterisk internal (192.168.254.254) --> SIP phone ( 192.168.254.104) > > I have no idea why asterisk is trying to send the outbound RTP stream to > itself. Removing the externhost and localnet settings doesn't help either. > Neither does setting "nat = yes", even in the example below.
nat = yes solved it in the example above. > > > SIP.conf: > externhost = <external hostname --> ddns.org> > canreinvite = nonat > localnet = 192.168.254.0/24 > ; nat = option is not set. > > Outgoing call to our sip provider ends up being setup like this: > > outbound RTP stream: > SIP phone (192.168.254.104) --> asterisk internal (192.168.254.254) > asterisk external (internet IP) --> SIP provider (internet IP) > > inbound RTP stream: > SIP provider (internet IP) --> asterisk external (internet IP) > asterisk internal (192.168.254.254) --> SIP phone (192.168.254.104) > > The inbound RTP stream goes well for +/- 1 second, then the SIP provider > responds to a re-invite sent by my asterisk box to send the trafic to > 192.168.254.104 (the SIP phone on my internal network). > > outbound RTP stream: > SIP phone (192.168.254.104) --> asterisk internal (192.168.254.254) > asterisk external (internet IP) --> SIP provider (internet IP) > > inbound RTP stream: > SIP provider (internet IP) --> SIP phone (192.168.254.104) > > I don't understand the logic of Asterisk sending the re-invite for inbound > RTP stream. I would be more logical if Asterisk would send an invite for the > outbound RTP stream: > > outbound RTP stream: > SIP phone (192.168.254.104) --> SIP provider (internet IP) > > inbound RTP stream: > SIP provider (internet IP) --> asterisk external (internet IP) > asterisk internal IP (192.168.254.254) --> SIP phone (192.168.254.104) > > Does the logic have anything to do with in which order the interfaces are > defined on the box? In my case, ETH0 = 192.168.254.254, ETH1 = internet IP. > > I can't find any configuration examples of my kind of setup, where a > dual-homed host running asterisk has one NIC on the Internet and one on the > internal (RFC1918 space) network. All examples I've bumped into have either > the asterisk box behind a NAT router ( i.e. it only has a RFC1918 > IP-address) or the asterisk box is on a real IP. > > with kind regards, > > Stefan _______________________________________________ --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
