you didn't by accident remove the internal ip range that your system resides on?
either that or your DNS is messed up... On Fri, Aug 5, 2011 at 3:21 PM, Max DiOrio <[email protected]> wrote: > Sending profile gives > domain-config failed > Sipxsupervisor-config failed > Alarm-config failed > Alarm-group failed > Alias failed > Validusets.XML failed > > Etc > > Michael Picher <[email protected]> wrote: > > > 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 > > > _______________________________________________ > 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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