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. 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
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