On 2/10/07, Luki <[EMAIL PROTECTED]> wrote:
Stefan, > When I have 2 SIP endpoints that both aren't configured with > "canreinvite=no" then I get no sound. The Sipura 3102 definitely works fine with canreinvite=yes and I never really had a problem with any of the Sipura devices in this respect, especially when there is no NAT involved. However, the default "Auto NetService Private IP Ranges:" includes 192.168.0.0-192.168.255.255, so your 192.168.254.0/24 network would be considered a LAN address by the 3102 and hence the traffic would go out the LAN interface (not WAN). Change this setting by removing this range. It's on the Admin > Advanced > LAN Setup tab. If that doesn't help, then you need to check what traffic is being sent. Since all devices are on the same internal network I assume they can see each other. You need to look at the Invite (and ReInvite) messages sent and received and see if the IP addresses for RTP listed there make sense. Then I suggest you use tcpdump to see what traffic is sent by each device, and where. If you have a switched network environment this will be a bit tricky as your * box won't see this traffic, so you may want to use a hub for this test (just temporarily) or if available set up port mirroring to sniff the traffic. Good luck and keep us posted.
Luki, I just configured the wlan phone and my eyebeam endpoints with canreinvite=yes (which should put the sipura out of the picture). Calling the wlan phone from eyebeam: no sound gets through. Putting a canreinvite=no in either one of the configurations (for the wlan01 or the eyebeam) forces the media through the asterisk and sound gets through. I'll get wireshark running on my laptop so I can post the SIP conversations here. regards, Stefan
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