They verified they are sending on 5080 in wireshark. I haven't had time to look and probably wwont until tomorrow at this point.
Tony Graziano <[email protected]> wrote: that would be correct if they were sending to the wrong port. On Aug 5, 2011 5:23 PM, "Max DiOrio" <[email protected]<mailto:[email protected]>> wrote: > No luck in setting up the trunk that way. They have said that they are now > getting a 404 from my server on the invite. > > Josh Patten <[email protected]<mailto:[email protected]>> wrote: > > > > EEK! > > You need to change the IP address in the sipXconfig interface under system >> > servers >> <name_of_server> >> Configure and then resend profiles. > > If that doesn't work you may need to stop sipxecs (service sipxecs stop) and > rerun the final setup portion (sipxecs-setup). This will regenerate your IP > and domain configs. > > You'll also need to change the IP DNS > > As for setting the "realm" to that "name", try setting the address of the SIP > trunk to 10.32.0.65 and set the outbound proxy address to 64.246.135.202 > > > On Fri, Aug 5, 2011 at 2:26 PM, Max DiOrio > <[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>> > wrote: > > What do you mean? I changed the internal LAN in pfsense. That's fine except > for LAN access to the internet. > > I then changed the IP of sipXecs server. Restarted the network services. Then > changed the IP of the server in the GUI. > > Michael Picher > <[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>> > wrote: > > > > 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]<mailto:[email protected]><mailto:[email protected]<mailto:[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]<mailto:[email protected]><mailto:[email protected]<mailto:[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]<mailto:[email protected]><mailto:[email protected]<mailto:[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]<mailto:[email protected]><mailto:[email protected]<mailto:[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]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>> > wrote: > > 5060 tcp/udp > 5080 udp > 30000-31000 udp > > Michael Picher > <[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>> > wrote: > > > > voip.ms<http://voip.ms><http://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]<mailto:[email protected]><mailto:[email protected]<mailto:[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]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> > > [[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>] > on behalf of Tony Graziano > [[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[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]<mailto:[email protected]><mailto:[email protected]<mailto:[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]<mailto:[email protected]><mailto:[email protected]<mailto:[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<http://static-72-10-215-161.albyny.csvoip.net><http://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]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> >>> [ >>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>] >>> on behalf of Tony Graziano [ >>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[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]<mailto:[email protected]><mailto:[email protected]<mailto:[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]<mailto:username%[email protected]><mailto:username%[email protected]<mailto: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]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> >>>> [ >>>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>] >>>> on behalf of Tony Graziano [ >>>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[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.com<http://bandwidth.com><http://bandwidth.com> template >>>> 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]<mailto:[email protected]><mailto:[email protected]<mailto:[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]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> >>>>> [ >>>>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>] >>>>> on behalf of Max DiOrio [ >>>>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[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]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> >>>>> [ >>>>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>] >>>>> on behalf of Michael Picher [ >>>>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[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]<mailto:[email protected]><mailto:[email protected]<mailto:[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]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> >>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Michael Picher >>>>> eZuce >>>>> Director of Technical Services >>>>> O.978-296-1005 X2015<tel:978-296-1005%20X2015> >>>>> M.207-956-0262<tel:207-956-0262> >>>>> @mpicher <http://twitter.com/mpicher> >>>>> www.ezuce.com<http://www.ezuce.com><http://www.ezuce.com> >>>>> >>>>> >>>>> _______________________________________________ >>>>> sipx-users mailing list >>>>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> >>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>> >>>> >>>> >>>> >>>> -- >>>> ====================== >>>> Tony Graziano, Manager >>>> Telephone: 434.984.8430<tel:434.984.8430> >>>> sip: >>>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> >>>> Fax: 434.326.5325<tel:434.326.5325> >>>> >>>> Email: >>>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> >>>> >>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>> Telephone: 434.984.8426<tel:434.984.8426> >>>> sip: >>>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[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]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> >>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> >>> >>> >>> >>> -- >>> ====================== >>> Tony Graziano, Manager >>> Telephone: 434.984.8430<tel:434.984.8430> >>> sip: >>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> >>> Fax: 434.326.5325<tel:434.326.5325> >>> >>> Email: >>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> >>> >>> LAN/Telephony/Security and Control Systems Helpdesk: >>> Telephone: 434.984.8426<tel:434.984.8426> >>> sip: >>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[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]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >> >> >> >> -- >> Michael Picher >> eZuce >> Director of Technical Services >> O.978-296-1005 X2015<tel:978-296-1005%20X2015> >> M.207-956-0262<tel:207-956-0262> >> @mpicher <http://twitter.com/mpicher> >> www.ezuce.com<http://www.ezuce.com><http://www.ezuce.com> > > _______________________________________________ > sipx-users mailing list > [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > -- > Michael Picher > eZuce > Director of Technical Services > O.978-296-1005 X2015<tel:978-296-1005%20X2015> > M.207-956-0262<tel:207-956-0262> > @mpicher <http://twitter.com/mpicher> > www.ezuce.com<http://www.ezuce.com><http://www.ezuce.com> > > > _______________________________________________ > sipx-users mailing list > [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > -- > Michael Picher > eZuce > Director of Technical Services > O.978-296-1005 X2015<tel:978-296-1005%20X2015> > M.207-956-0262<tel:207-956-0262> > @mpicher <http://twitter.com/mpicher> > www.ezuce.com<http://www.ezuce.com><http://www.ezuce.com> > > > _______________________________________________ > sipx-users mailing list > [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > -- > Michael Picher > eZuce > Director of Technical Services > O.978-296-1005 X2015<tel:978-296-1005%20X2015> > M.207-956-0262<tel:207-956-0262> > @mpicher <http://twitter.com/mpicher> > www.ezuce.com<http://www.ezuce.com><http://www.ezuce.com> > > > _______________________________________________ > sipx-users mailing list > [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > -- > Michael Picher > eZuce > Director of Technical Services > O.978-296-1005 X2015<tel:978-296-1005%20X2015> > M.207-956-0262<tel:207-956-0262> > @mpicher <http://twitter.com/mpicher> > www.ezuce.com<http://www.ezuce.com><http://www.ezuce.com> > > > _______________________________________________ > sipx-users mailing list > [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > -- > Josh Patten > eZuce > Solutions Architect > O.978-296-1005 X2050 > M.979-574-5699
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