BTW, I have since disabled the "sip-voip" plugin and re-enabled the inbound rules for sip/rtp on my firewall. I have noticed that on at least two occasions, I would receive a call via my sip provider and I could not hear the calling party (nor could they hear me). That tells me that the dynamic rules for rtp with the plugin were not always working well. With the last incident, I verified in my log that the rtp packets were in fact being blocked, see below:
Mar 16 13:12:56 pbx user.info kernel: AIF:UNPRIV UDP packet: IN=eth0 OUT= MAC=00:00:24:cc:11:68:00:26:62:21:ea:4b:08:00 SRC=66.241.96.96 DST=192.168.1.201 LEN=92 TOS=0x00 PREC=0x00 TTL=54 ID=0 DF PROTO=UDP SPT=14345 DPT=10099 LEN=72 Mar 16 13:13:01 pbx user.info kernel: AIF:UNPRIV UDP packet: IN=eth0 OUT= MAC=00:00:24:cc:11:68:00:26:62:21:ea:4b:08:00 SRC=66.241.96.96 DST=192.168.1.201 LEN=92 TOS=0x00 PREC=0x00 TTL=54 ID=0 DF PROTO=UDP SPT=14345 DPT=10099 LEN=72 No big deal using static rules in Arno, just thought someone might be interested in knowing about this. Also, if someone disagrees with my assessment, I'd like to know that as well! FYI, I'm running 0.7.7 on a Soekris Net 5501. -tm ------------------------------------------------------------------------------ Colocation vs. Managed Hosting A question and answer guide to determining the best fit for your organization - today and in the future. http://p.sf.net/sfu/internap-sfd2d _______________________________________________ Astlinux-users mailing list Astlinux-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pay...@krisk.org.
