[asterisk-users] canreinvite problems

[Stefan van der Eijk]

On 2/10/07, Luki <lugosoft at gmail.com> 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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