1. Run local DNS for your private network. 2. Port forward 8443 from your DNS provider to your personal router. Make sure your router relays it to your local sipx installation.
Everything will function normally inside for internal calls. management can be done from the outside. There is really no need to do public SRV records since you haven't indicated you have a FXO gateway internally (not that it matters), or a method to traverse your firewall/router for remote SIP users or ITSP trunking. NAT traversal for SIP is not a simple matter handled by a home broadband router, nut running local DNS will make sipx run much more reliably, and makie registrations less problematic. Tony >>> Roger Toennis <[EMAIL PROTECTED]> 07/13/08 17:58 PM >>> Hi Scott, Thanks for email. Registration was going fine but when I made calls on X Lit sofphones with the calls failed. I have sipx set with no local DNS and instead nameservers set to what comcast has on their network in my area. However I just discovered that if I put in the local IP of my sipx on my lan into the domain field for X Lite I can make calls. I have godaddy set to direct sipx.<my domain>.com to my external IP on my router and my router is port forwarding 8443 so I can manage the sipx remotely. I guess maybe what I need is to set up an SRV rules on godaddy for the domain for the SIP traffic so when the X Lite phones send sip traffic to "sipx.mydomain.com" it gets forwarded to my router IP. Router then needs to port forward that traffic on the known ports on sipx?? Does that make sense? And if so what are the specifics I need to setup for port forwarding 5060, etca al. TCP? UDP? both? RTP media from external? what port ranges? ??? Thanks, Roger On Jul 13, 2008, at 3:09 PM, Scott Lawrence wrote: > > On Sun, 2008-07-13 at 14:33 -0600, Roger Toennis wrote: >> Centos 3.10.1 single disk install >> on home LAN behind firewall >> No local DNS on SIPX...... >> >> So I was able to get 2 users configured and register those users with >> XLite softphones from PCs on my local LAN. >> However when I try making calls between the registered extension the >> calls failed with "user unavailable". >> >> What am I missing that I havn't done? > > http://sipx-wiki.calivia.com/index.php/ > SipX_Phone_Registration_Troubleshooting > > -- > Scott Lawrence tel:+1.781.229.0533;ext=162 or > sip:[EMAIL PROTECTED] > sipXecs project coordinator - SIPfoundry http:// > www.sipfoundry.org/sipXecs > CTO, Voice Solutions - Bluesocket Inc. http://www.bluesocket.com/ > http://www.pingtel.com/ > > -- Roger Toennis President/CEO Liquid Media Software LLC 303-522-5477 "Making the simple complicated is commonplace. Making the complicated simple, that's creativity" - Charles Mingus CONFIDENTIALITY NOTICE: This e-mail, and any attachments thereto, is intended only for use by the addressee(s) named herein and may contain privileged and/or confidential information. If you are not the intended recipient of this e-mail, you are hereby notified that any dissemination, distribution or copying of this e-mail, and any attachments thereto, is strictly prohibited. _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
