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.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]> 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
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