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
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