Hi Tom, Thanks for documenting the issues with the sip-voip plugin. I've come to the same conclusion over the years but never documented it.
Good advice is to leave the sip-voip plugin disabled. Lonnie On Mar 16, 2011, at 12:43 PM, Tom Mazzotta wrote: > 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.
