> When I take a phone and plug it into a wireless remote > network connection, the following keeps showing up in sipXproxy.log. > > "2010-03-25T19:48:50.052314Z":3099:INCOMING:INFO:uc.domain.com > :SipClientUdp-8:B78FCB90:SipXProxy:"Read SIP > message:\n----Remote Host:xxx.xxx.xxx.215---- Port: > 5060----\nSIP/2.0 404 Not Found\r\nTo: > sip:anonym...@anonymous.invalid;tag=48c90a4ff339852i0\r\nFrom: > \"SipXecs Keepalive\"<sip:an > onym...@anonymous.invalid>;tag=30543f3483e1cb11ecb40866edd3295 > b-reniatniamtan\r\nCall-Id: > f88dfabce84b6a2787ef024a7dbe8749-reniatniamtan\r\nCseq: > 1 OPTIONS\r\nVia: SIP/2.0/UDP > xxx.xxx.xxx.75:5060;branch=z9hG4bK-234fc22f2nnthda-reniatniamt > an\r\nServer: Linksys/SPA941-5.1.8\r\nContent-Length: 0\r\n\r\n
This is a 'normal error'. The sipXproxy will periodically (every 20 secs.) send a quasi-bogus OPTIONS message to every phone that is behind a remote NAT. That OPTIONS message is crafted with anonymous To and From fields. The purpose of that message is to maintain the firewall pinhole open at the remote router without causing the remote phone to spring into action (like stating to ring). Different phone models respond differently to this OPTIONS message. Some '200 OK' it, some '404 Not Found' it and some others generate other kinds of error response. The exact nature of the response does not matter. What matter is that the phone does respond with something to maintain the pinhole. So, what I conclude from looking at this log entry is that your phone was successfully recognized as a remote user behind a NAT otherwise the proxy would not have sent it an OPTIONS message in the first place. > > When I take the exact same phone, change nothing on it and > connect it locally, then the user information shows up as it should. > All remote users are showing up as anonymous since I started > testing a pfsense box connection to sipx. > > All remote users can make outgoing calls but cannot receive > calls. Depending on settings changed on sipx such as Internet > calling for example, all we get is a busy signal. Stay away from Internet calling if you can avoid it. Please visit http://wiki.sipfoundry.org/display/xecsuserV4r0/Remote+User+NAT+Traversal for general instructions on how to setup remote workers on sipXecs. > > Google searching says that this is simply a keep-alive > function since it's just a device reconnecting to stay > connected but since incoming calls aren't working, I wanted > to confirm this. I messed something up when putting the > pfsense box up and now can't figure out what I did so am > looking for leads :). Check out "Problem #2" in link above and check that the IP addresses that show up in sipXconfig's registrations diagnostics page are as you would expect. If not, you have router config issues that you need to work out. > > The mix of hardware is an audiocodes 2K on the LAN side and > remote users over pfsense at the moment and sipx 4.0.4. > > Mike > > _______________________________________________ > sipx-users mailing list sipx-users@list.sipfoundry.org 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 sipx-users@list.sipfoundry.org 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/
