probably belongs in the wiki... On Thu, Mar 22, 2012 at 2:46 PM, Robert Schroeder < [email protected]> wrote:
> I agree completely. I will be posting two different firewall > configurations that I have testing. The firewalls tested are (WatchGuard > XTM 510 & Kerio 7.3.0).**** > > ** ** > > Is it Ok with the forum to insert pictures into the postings? **** > > ** ** > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Tony Graziano > *Sent:* Thursday, March 22, 2012 2:02 PM > *To:* Discussion list for users of sipXecs software > *Subject:* Re: [sipx-users] VoIP.ms Setup Experience**** > > ** ** > > The only challenge here is the firewall configuration. Perhaps if you > posted your particular firewall configuration others can learn from it.*** > * > > On Mar 22, 2012 1:18 PM, "Robert Schroeder" < > [email protected]> wrote:**** > > Well today I undertook the setup challenge of performing an ITSP setup on > one of my sipXecs (Test & Tears) servers. I must say the experience was a > bit challenging at first with just plain trying to figure out the process. > I have listed a link below that was a little help in the process. The setup > of the SIP Trunk Gateway was straight forward and the setup of the VoIP.ms > account was a little stressful. The most challenging part of the process > was configuring the 1:1 NAT or Full Clone NAT on the WatchGuard firewall. > In the end it always comes down to the simply statement of “Oh Now I See”! > **** > > **** > > I was in the end (2 -3 Hours) able to get things configured and working so > that the SIP SBC Trunk Gateway is listed as “Authenticated” under the > Diagnostics Tab – SIP SBC Trunk Gateway area. While my testing thus far has > gone very well I am in no way ready to go “Only” ITSP and lay aside my > assurances with traditional SIP gateways connected to PRI or POTS circuits. > I am starting to believe though.**** > > **** > > Areas Tested:**** > > **** > > Outbound Dialing: Good Results Thus Far**** > > Inbound Call Routing from VoIP.MS: Good Results Thus Far**** > > VoIP.MS DID routing to sipXecs Accounts: Good Results Thus Far**** > > MOH Testing: Good Results Thus Far**** > > **** > > The only problem that I sometimes have is when I first dial a number. The > softphone shows that the call failed however my cell rings a couple of > times. When I try to do this a second time the issue does not occur again. > If I wait a few minutes and try again the issue does not show up. If I wait > a long while then the problems occurs again. I am not sure if the gateway > connection to voip.ms is interrupted if no calls are made for a while. My > thoughts are Session Timer Interval or Registration Interval in the SIP > Trunk Gateway configuration. Perhaps it could even be a Firewall Issue. > *Thoughts > anyone!***** > > **** > > **** > > I will be posting later a simple setup guide for (VoIP.ms) with sipXecs. > Those that are *more* informed, experienced and intelligent than I, > please be gentle. I would like to in the end have a good instruction set > for the user community when connecting to VoIP.ms. The suggestions and > corrections made by those that are far smarter than I can only be > considered a greater asset. **** > > **** > > http://blog.myitdepartment.net/?p=191**** > > **** > > **** > > Thanks Everyone,**** > > **** > > Rob**** > > ** ** > ------------------------------ > > > NOTICE: This electronic mail message and any content within it are intended > exclusively for the individual(s) or > > entities to which it is addressed. The message, together with any attachments > and all other content, may contain > > confidential and/or privileged information. Any unauthorized review, use, > print, save, copy, disclosure or distribution > > is strictly prohibited. If you have received this message in error, please > immediately advise the sender by reply email > and delete all copies.**** > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/**** > > ** ** > > LAN/Telephony/Security and Control Systems Helpdesk:**** > > Telephone: 434.984.8426**** > > sip: [email protected]**** > > ** ** > > Helpdesk Customers: http://myhelp.myitdepartment.net**** > > Blog: http://blog.myitdepartment.net**** > > > ------------------------------ > > NOTICE: This electronic mail message and any content within it are intended > exclusively for the individual(s) or > > entities to which it is addressed. The message, together with any attachments > and all other content, may contain > > confidential and/or privileged information. Any unauthorized review, use, > print, save, copy, disclosure or distribution > > is strictly prohibited. If you have received this message in error, please > immediately advise the sender by reply email > and delete all copies. > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square**** Suite 201**** Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher <http://twitter.com/mpicher> www.ezuce.com ------------------------------------------------------------------------------------------------------------ There are 10 kinds of people in the world, those who understand binary and those who don't.
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
