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Title: How to get SIP to work on Community Editon 5.5.1 Post by: bdmeyer on Wednesday 01 August 2012, 11:30:37 am I understand there is no more SIP proxy.
I added this to the outgoing rules: Code: 6c:33:a9:11:f1:a1 RED TCP+UDP/5060 TCP+UDP/5070 ALLOW SIP I am pretty certain I only need UDP. When i enable outbound rules, I get dial tone but no voice. Eth1 is my DMZ. Nothing else is on it. i tried the same rules on the Green Interface. Same and results. With TCPDump I see this when I dial out: Code: tcpdump -i eth1 port 5060 or port 5070 -nnv iptables -L (grep for MAC) Code: ACCEPT tcp -- anywhere anywhere tcp dpt:sip MAC 6c:33:a9:11:f1:a1 tailing the firewall log I see this though: Code: INPUTFW:ACCEPT:3:l3 IN=br1 OUT= MAC=00:1b:2f:35:82:72:6c:33:a9:11:f1:a1:08:00 SRC=192.168.2.150 DST=192.168.2.1 LEN=60 TOS=00 PREC=0x00 TTL=255 ID=1435 PROTO=ICMP TYPE=8 CODE=0 ID=0 SEQ=0 MARK=3000 Is my device using non standard ports, or is the ports shown above on these dropped packets normal for a SIP phone? Thank you for any guidance. -= Bruce Title: Re: How to get SIP to work on Community Editon 5.5.1 Post by: trymes on Friday 03 August 2012, 12:05:44 am SIP communicates using more than one port. One is for signalling, usually 5060, and the other is for the RTP traffic, which is the voice/video/etc.
Generally, you need to allow for both port 5060 and a range of ports for RTP. When using asterisk, for example, I generally forward 5060 and 10,000-20,000, but those numbers vary depending on the settings and type of your SIP server. Tom Title: Re: How to get SIP to work on Community Editon 5.5.1 Post by: bdmeyer on Friday 03 August 2012, 12:32:03 am Thanks for the reply Tom,
I'll figure what the port range is, and give it a try. I ended up in the mean time just setting a rule for the MAC address of the device and then UDP port ANY which works, but is sloppy. -= Bruce |