Ok, will follow these suggestions also and update. Thanks.
On Sat, 7 Nov 2009 10:22:43 -0500, Tony Graziano wrote: > pfsense do not have alg, though I would: > > 1. Delete the siproxd package it has. > 2. Make sure AON is enabled (the sipx wiki has some details about static > NAT and how to enable it in sipx. > > > I also posted a sample config for pfsense (xml document) you can modify and > restore to your pfsense box as well as a traffic shaper wizard that > includes sipxbridge/sipxecs. I think I built a pfsense box and did these > two steps in under thirty minutes. Your deployment is a little different > with vlans, but I think this will help you in any case. > > > blog.myitdepartment.net. > > On Sat, Nov 7, 2009 at 10:16 AM, [email protected] <[email protected]> > wrote: >>> Each box can have their own SIP domain and for the purposes of testing >>> and >>> characterization, remote users could have a line on either (or even >>> both). >>> >> >> My hope was to have fail over ahead of the sipx box. The DNS load >> balancing with opensbc >> is pretty cool, I can watch connections flip flopping across both sides. >> I have also wondered if that could be the source of the problem as well. >> A fixed phone >> seems to stick to what ever sbc is registered through until the session >> dies and it >> reconnects, then it can hit either one again but seems to stick fine >> otherwise. >> >> When I watch say a soft phone trying to connect, unless it connects >> immediately (registers >> I mean), then I can see it flip flopping between the sbc's. That has made >> me wonder many >> times if it's possible that sipx could be having trouble sending data >> back out to the >> proper sbc for some reason, perhaps failing the registrations? >> >> I've also seen where even a fixed phone is (remote phone, registered over >> Internet to >> sbc's) connected, seems to be working fine. The person will pick up the >> phone to make a >> call and has one way audio. This happened the other day when we were >> testing conference >> calls. Everything worked, then suddenly, the remote sip phone had one way >> audio the next >> call. >> >>> enable remote NAT traversal on sipXecs are described in >>> http://sipx- >>> wiki.calivia.com/index.php/Configuring_remote_workers_cheatsheet. >>> >> >> Alright, I'll get to that shortly. >> >>> Once you get that going, perhaps you can share your results on >>> this mailing list (under a new thread) and I will work with you to make >>> it work reliably or at least understand exactly why your setup is not >>> >> >> I'll start a new thread when done. Here's what I have so far and I'll >> post this part in >> that new thread as well. >> >> Built a PfSense server and a new sipx 4.0.2. >> Searching google, the pfsense wiki and forums, it appears that PfSense >> does NOT have ALG. >> So if someone disagrees with using this firewall, I'll use something >> else. In the >> meantime; >> >> WAN Router >> | >> Switch >> | >> Eth1-Public IP x.x.x.x >> PfSense >> Eth0-LAN IP (VLAN) (Passing SIP/RTP only. All other services over >> different connection) >> | >> Switch >> | >> BladeCenter Blade Running ESX - One Guest only, SipX 4.0.2 >> >> The final setup intention is one server on each WAN for fail >> over/redundancy. >> I don't really want to post any real IPs so if there will be testing to >> be done, perhaps I >> can send some info off list and we can post results if that's ok. >> >> Mike >> >> >> _______________________________________________ >> sipx-users mailing list [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users >> Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users >> sipXecs IP PBX -- http://www.sipfoundry.org/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
