5060 tcp/udp 5080 udp 30000-31000 udp Michael Picher <[email protected]> wrote:
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]>> 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]> [[email protected]<mailto:[email protected]>] on behalf of Tony Graziano [[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]>> 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]>> 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> >> (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]> >> [ >> [email protected]<mailto:[email protected]>] >> on behalf of Tony Graziano [ >> [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]>> 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]> >>> >>> 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]> >>> [ >>> [email protected]<mailto:[email protected]>] >>> on behalf of Tony Graziano [ >>> [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> 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]>> 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]> >>>> [ >>>> [email protected]<mailto:[email protected]>] >>>> on behalf of Max DiOrio [ >>>> [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]> >>>> [ >>>> [email protected]<mailto:[email protected]>] >>>> on behalf of Michael Picher [ >>>> [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]>>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]> >>>>> 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> >>>> >>>> >>>> _______________________________________________ >>>> sipx-users mailing list >>>> [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]> >>> Fax: 434.326.5325<tel:434.326.5325> >>> >>> Email: [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]> >>> >>> 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]> >>> 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]> >> Fax: 434.326.5325<tel:434.326.5325> >> >> Email: [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]> >> >> 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]> >> 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> _______________________________________________ sipx-users mailing list [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 M.207-956-0262 @mpicher <http://twitter.com/mpicher> www.ezuce.com<http://www.ezuce.com>
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