SIP appears to be partially blocked - IM/P2P/IDS

Hi,
since I upgraded to v7.180 my voip phone won't register. I have removed all SIP helpers and have the voip phone -> any-> any-> allow with logging.

The voip phone shows a high number of packets sent, but a low number received. So what is the IM/P2P/IDS rules that cause the packets to vanish?

Also I have this strange entry in my PF log, see the attached jpeg for the "auto generated filter". What caused it to autogenerate?

Ian M

0 Mike H over 17 years ago

Autogenerated are from rules like DNAT/SNAT etc where there is a checkbox (checked) to auto create the packefilter rule.

Edit - Just to clarify. I created a DNAT rule for some peer torrent traffic with the create packetfilter checkbox. In the live log all the peer traffic went from red/dropped to grey/auto created. I then unchecked the create packet filter, made my own rule without logging yet the logging continued until I rebooted the box. I have tried this with other new (not existing) DNAT/SNAT rules and have not been able to get them to log like that. Perhaps I am completely wrong or perhaps that logging ability is horribly broken.
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0 -INSOMNIAC- over 17 years ago

hey.
please take a look @ https://community.sophos.com/products/unified-threat-management/astaroorg/f/92/t/66554

seems like this could also fit to your problems.

regards
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0 RFCat_vk_01 over 17 years ago in reply to -INSOMNIAC-

Hi,
it goes part the way, that was why I started a new thread. I have not created any new NAT rules for a while and only changed winny from blocked to alert.

I still don't have any definite idea about which rule is blocking SIP, but suspect it could be the skype rule. Something to try tonight when I get home from work. What ever rule it is seems to override the SIP helper function which I have disabled for the moment.

Ian M
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0 RFCat_vk_01 over 17 years ago in reply to RFCat_vk_01

I removed IM/P2P and IDS, maybe not for long enough and still the packets did not reach the VOIP ATA. Lots of green packets in the PF all outgoing, none incoming.

Ian M
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0 RFCat_vk_01 over 17 years ago in reply to RFCat_vk_01

Would appear that while I was away on holidays my VoIP provider changed their network without advising anyone.
A lot of reading forums and guess work has restored the VoIP service simplified the network configurations and services.

The default SIP in ASG is TCP where a number of VoIP operators use UDP. Created a generic TCP/UDP and added that to the VoIP group. Still need PF rule to allow the RTP packets out.

The only thing left to resolve is why some of the returned packets use the firewall address and get dropped. There doesn't appear to be any affect on the voice quality.

Ian M[[:)]][[:)]]
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