Make sure your NAT settings has your external IP listed static and that you have re-sent your server profile...
Usually you want to 'skinny' down that IP range to be just your internal IP's. Mike On Fri, Aug 5, 2011 at 3:10 PM, Max DiOrio <[email protected]> wrote: > I just changed my internal IP range. SipXecs is acting strangely now. > Trying to view trunk registrations I'm getting sipxbridge xml-rpc error. > > Michael Picher <[email protected]> wrote: > > > ok, if you are running pfsense, capture on the outside interface and make > sure that your calls are coming from the metaswitch and coming to port > 5080udp on the outside ip of your firewall. > > what subnets are in your 'internet calling' page on sipxecs? > > On Fri, Aug 5, 2011 at 2:53 PM, Max DiOrio <[email protected]> wrote: > >> 5060 tcp/udp >> 5080 udp >> 30000-31000 udp >> >> Michael Picher <[email protected]> wrote: >> >> >> voip.ms doesn't need inbound ports mapped... >> >> metaswitch will need to have ports mapped.... >> >> let us know what you have mapped for ports inbound. >> >> On Fri, Aug 5, 2011 at 2:37 PM, Max DiOrio <[email protected]> wrote: >> >>> Could it possibly be something in my pfsense firewall affecting >>> anything? >>> >>> >>> >>> I followed the template the Tony put out there on his website. Voip.ms >>> is working fine, which leads me to believe that no, the firewall is fine. >>> >>> >>> >>> Is there a way to quickly shut off the firewall in pfsense to test? >>> >>> >>> >>> ------------------------------ >>> >>> *From:* [email protected] [ >>> [email protected]] on behalf of Tony Graziano [ >>> [email protected]] >>> *Sent:* Friday, August 05, 2011 2:32 PM >>> >>> *To:* Discussion list for users of sipXecs software >>> *Subject:* Re: [sipx-users] Problems with Metaswitch at ITSP >>> >>> just because they can do what they please doesn't mean they should do >>> what they please. >>> >>> to me that would present a horrible security and routing situation. >>> On Aug 5, 2011 2:30 PM, "Michael Picher" <[email protected]> wrote: >>> > no audio = media relay problem... again, check internet communications >>> > settings page... >>> > >>> > On Fri, Aug 5, 2011 at 2:25 PM, Max DiOrio <[email protected]> >>> wrote: >>> > >>> >> That's what I was thinking. I didn't even think about them routing it >>> >> differently. >>> >> >>> >> >>> >> >>> >> Both my data provider and ITSP are the same provider, so I guess they >>> can >>> >> do what they please. >>> >> >>> >> >>> >> >>> >> [root@phones sipxpbx]# tracert 10.32.0.65 >>> >> traceroute to 10.32.0.65 (10.32.0.65), 30 hops max, 40 byte packets >>> >> 1 10.0.0.82 (10.0.0.82) 0.086 ms 0.077 ms 0.112 ms >>> >> 2 static-72-10-215-161.albyny.csvoip.net (72.10.215.161) 14.401 ms >>> >> 14.405 ms 14.406 ms >>> >> 3 * * * >>> >> >>> >> 4 * * * >>> >> >>> >> 5 * * * >>> >> >>> >> 6 * * * >>> >> >>> >> 7 * * * >>> >> >>> >> etc. >>> >> >>> >> >>> >> >>> >> I can now get one of me 3 DID's to ring inbound, but have no audio on >>> it. >>> >> The other two still won't go through. >>> >> >>> >> >>> >> ------------------------------ >>> >> *From:* [email protected] [ >>> >> [email protected]] on behalf of Tony Graziano [ >>> >> [email protected]] >>> >> *Sent:* Friday, August 05, 2011 2:18 PM >>> >> >>> >> *To:* Discussion list for users of sipXecs software >>> >> *Subject:* Re: [sipx-users] Problems with Metaswitch at ITSP >>> >> >>> >> I don't think that will legally route. they are smoking some serious >>> you >>> >> know what. Can you get them to talk to you intelligently without them >>> >> giggling and needing munchies? >>> >> >>> >> Domain name or not, its dotted quad, it's an ip address. There is no >>> >> registered domain name of that one the internet. So they just want you >>> to >>> >> use an IP address instead. They can call it whatever they want, it's >>> still >>> >> an IP ADDRESS until we get wasted and want to call it something it's >>> not. >>> >> Kinda like calling the current US deficit a temporary bookkeeping >>> error. >>> >> Not. >>> >> >>> >> Now, can you traceroute to them over your Internet connection and get >>> to >>> >> their network at that IP? I hate when isp's. break stuff like that. >>> >> >>> >> this means it will route over their network to you but not from any >>> >> internet connection. >>> >> >>> >> >>> >> >>> >> On Fri, Aug 5, 2011 at 2:00 PM, Max DiOrio <[email protected]> >>> wrote: >>> >> >>> >>> Here's what they gave me: >>> >>> >>> >>> >>> >>> >>> >>> SIP Authentication >>> >>> u: didnumber >>> >>> p: password >>> >>> >>> >>> SIP Proxy: 64.246.135.202 >>> >>> Domain Name - 10.32.0.65(yes, it's a 'name') >>> >>> >>> >>> In Asterisk, these are the settings that worked: >>> >>> >>> >>> >>> >>> >>> >>> Trunk Name: Cornerstone >>> >>> Outgoing Peer Details: >>> >>> Host=64.246.135.202 >>> >>> Username=providedbyCStel >>> >>> Secret=providedbyCSTel >>> >>> Type=friend >>> >>> Insecure=very >>> >>> Realm=10.32.0.65 >>> >>> Registration String: username:[email protected] >>> >>> >>> >>> I am registering and and able to place calls, just not receiving any. >>> >>> If I put the realm in the ITSP address, it won't register. >>> >>> ------------------------------ >>> >>> *From:* [email protected] [ >>> >>> [email protected]] on behalf of Tony Graziano [ >>> >>> [email protected]] >>> >>> *Sent:* Friday, August 05, 2011 1:42 PM >>> >>> >>> >>> *To:* Discussion list for users of sipXecs software >>> >>> *Subject:* Re: [sipx-users] Problems with Metaswitch at ITSP >>> >>> >>> >>> This really is not that hard. It's EASIER than with ASTERISK. >>> >>> >>> >>> It probably needs to go in as the gateway address and the ITSP server >>> >>> address. I don't know why this matters. You are gistering and sending >>> them >>> >>> calls? >>> >>> >>> >>> Do you mean the ITSP is sending you calls from another IP address >>> >>> altogether different from the above? If so, that really sucks and >>> makes me >>> >>> think you will see more compatibility issues. Who is the ITSP? If >>> this IS >>> >>> the case, you would create a sip trunk using the bandwidth.comtemplate >>> >>> and just put the IP in both the places mentioned above so it gets >>> included >>> >>> in an ACL for sipxbridge to know it allowed and treat it as a trunk >>> call. >>> >>> >>> >>> On Fri, Aug 5, 2011 at 1:38 PM, Max DiOrio <[email protected]> >>> wrote: >>> >>> >>> >>>> The ITSP gave me a domain name to use that's an IP address. In >>> >>>> Asterisk, this was put in as a realm in the trunk config. Where >>> would it go >>> >>>> in sipXecs, or is it needed? >>> >>>> >>> >>>> >>> >>>> ------------------------------ >>> >>>> *From:* [email protected] [ >>> >>>> [email protected]] on behalf of Max DiOrio [ >>> >>>> [email protected]] >>> >>>> *Sent:* Friday, August 05, 2011 1:23 PM >>> >>>> >>> >>>> *To:* Discussion list for users of sipXecs software >>> >>>> *Subject:* Re: [sipx-users] Problems with Metaswitch at ITSP >>> >>>> >>> >>>> I just forwarded them the info, hopefully it will help them out. >>> >>>> >>> >>>> >>> >>>> >>> >>>> I must say that sipXecs has a bunch of really helpful people who >>> really >>> >>>> know their stuff. >>> >>>> >>> >>>> >>> >>>> >>> >>>> sipXecs has been rock solid in my tesing so far. >>> >>>> >>> >>>> >>> >>>> >>> >>>> ------------------------------ >>> >>>> >>> >>>> *From:* [email protected] [ >>> >>>> [email protected]] on behalf of Michael Picher >>> [ >>> >>>> [email protected]] >>> >>>> *Sent:* Friday, August 05, 2011 1:16 PM >>> >>>> *To:* Discussion list for users of sipXecs software >>> >>>> *Subject:* Re: [sipx-users] Problems with Metaswitch at ITSP >>> >>>> >>> >>>> Usually with the Metaswitch they'll need to setup something to send >>> to >>> >>>> you on port 5080 udp. You then need to mak 5080 outside to 5080 >>> inside >>> >>>> (sipxbridge). >>> >>>> >>> >>>> Mike >>> >>>> >>> >>>> On Fri, Aug 5, 2011 at 1:02 PM, Max DiOrio <[email protected] >>> >wrote: >>> >>>> >>> >>>>> I'm just getting sipXecs set up with my ITSP, a local provider that >>> >>>>> uses a metaswitch at their end. >>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> I have the trunk registered with them and I can place outbound >>> calls >>> >>>>> without any issues. However inbound calls aren't even touching my >>> server, >>> >>>>> and I'm seeing nothing blocked in my firewall. >>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> VOIP.ms traffic works fine both directions. >>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> Does anyone have any similar metaswitch experience or know where I >>> can >>> >>>>> point my ITSP. They did a wireshark of their traffic and they >>> aren't seeing >>> >>>>> any problems. Their primary tech supoprt referred it to their >>> switch >>> >>>>> support. >>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> I'm just hoping that someone out there can lend some knowledge >>> since I'm >>> >>>>> down at this point. >>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> _______________________________________________ >>> >>>>> sipx-users mailing list >>> >>>>> [email protected] >>> >>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >>>>> >>> >>>> >>> >>>> >>> >>>> >>> >>>> -- >>> >>>> Michael Picher >>> >>>> eZuce >>> >>>> Director of Technical Services >>> >>>> O.978-296-1005 X2015 >>> >>>> M.207-956-0262 >>> >>>> @mpicher <http://twitter.com/mpicher> >>> >>>> www.ezuce.com >>> >>>> >>> >>>> >>> >>>> _______________________________________________ >>> >>>> sipx-users mailing list >>> >>>> [email protected] >>> >>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >>>> >>> >>> >>> >>> >>> >>> >>> >>> -- >>> >>> ====================== >>> >>> Tony Graziano, Manager >>> >>> Telephone: 434.984.8430 >>> >>> sip: [email protected] >>> >>> Fax: 434.326.5325 >>> >>> >>> >>> Email: [email protected] >>> >>> >>> >>> LAN/Telephony/Security and Control Systems Helpdesk: >>> >>> Telephone: 434.984.8426 >>> >>> sip: [email protected] >>> >>> >>> >>> Helpdesk Contract Customers: >>> >>> http://support.myitdepartment.net >>> >>> >>> >>> <http://support.myitdepartment.net>Blog: >>> >>> http://blog.myitdepartment.net >>> >>> >>> >>> Linked-In Profile: >>> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>> >>> >>> >>> Ask about our voip fax services! >>> >>> >>> >>> >>> >>> _______________________________________________ >>> >>> sipx-users mailing list >>> >>> [email protected] >>> >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >>> >>> >> >>> >> >>> >> >>> >> -- >>> >> ====================== >>> >> Tony Graziano, Manager >>> >> Telephone: 434.984.8430 >>> >> sip: [email protected] >>> >> Fax: 434.326.5325 >>> >> >>> >> Email: [email protected] >>> >> >>> >> LAN/Telephony/Security and Control Systems Helpdesk: >>> >> Telephone: 434.984.8426 >>> >> sip: [email protected] >>> >> >>> >> Helpdesk Contract Customers: >>> >> http://support.myitdepartment.net >>> >> >>> >> <http://support.myitdepartment.net>Blog: >>> >> http://blog.myitdepartment.net >>> >> >>> >> Linked-In Profile: >>> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>> >> >>> >> Ask about our voip fax services! >>> >> >>> >> >>> >> _______________________________________________ >>> >> sipx-users mailing list >>> >> [email protected] >>> >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >> >>> > >>> > >>> > >>> > -- >>> > Michael Picher >>> > eZuce >>> > Director of Technical Services >>> > O.978-296-1005 X2015 >>> > M.207-956-0262 >>> > @mpicher <http://twitter.com/mpicher> >>> > www.ezuce.com >>> >>> _______________________________________________ >>> sipx-users mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >> >> >> >> -- >> Michael Picher >> eZuce >> Director of Technical Services >> O.978-296-1005 X2015 >> M.207-956-0262 >> @mpicher <http://twitter.com/mpicher> >> www.ezuce.com >> >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > > > > -- > Michael Picher > eZuce > Director of Technical Services > O.978-296-1005 X2015 > M.207-956-0262 > @mpicher <http://twitter.com/mpicher> > www.ezuce.com > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- Michael Picher eZuce Director of Technical Services O.978-296-1005 X2015 M.207-956-0262 @mpicher <http://twitter.com/mpicher> www.ezuce.com
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