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
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