Hi there,
after years of using Astaro (since V3.1) I have now a problem which
I have not solved yet (Version 9.204-20).
End of this week our phone line will be switched to IP (VoIP). Therefor I prepare the UTM for this day and want to test it with a sipgate.de account. The internal PBX is a Starface. VoIP Support is enabled and all Sipgate servers and the IP of the local PBX is entered correctely at the VoIP Support
page. (Strict mode)
Internet access with PPPoE from the UTM.
With wireshark I inspected the packets on the pbx going to the UTM:
registering and dialing is working as expected. Phones are ringing but no tone is going in any direction!
The Ports stated in the SIP-Session description are used by the pbx for outgoing RTP. But UTM is blocking them all. (Default DROP)
Strange thing: Outgoing packets are marked as 'UDP' - Default DROP ; Incoming packets are marked as 'RTP' - Default DROP. Wireshark shows the outgoing packets as RTP and not as UDP... Is the helper not recognising the RTP packets correctely?
If I enter a Firewall rule (PBX -> AllSipgateServer -> Any) everything works fine - but I do not want to have an 'ANY' Rule in the list...
What's going wrong here?
Thanks for your help in advance
Marc
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