My Internet Calling settings were in the list of internal settings (10.0.0.0/8) I changed it to 10.0.0.0/27, my actual network CIDR subnet. Restarted the services, and still no luck.
They claim that they did change the signaling to port 5080 but it's still failing. ________________________________ From: [email protected] [[email protected]] on behalf of Michael Picher [[email protected]] Sent: Friday, August 05, 2011 2:24 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Problems with Metaswitch at ITSP You'll need to tweak your Internet Calling settings and make sure that 10.x.x.x address is not in your list of internal addresses. Likewise, you didn't touch on port 5080 udp. This must be the port that the metaswitch sends to you on and you need to map that back in to the sipXecs server. Ignore this at your own peril. On Fri, Aug 5, 2011 at 2:18 PM, Tony Graziano <[email protected]<mailto:[email protected]>> wrote: 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 M.207-956-0262 @mpicher <http://twitter.com/mpicher> www.ezuce.com<http://www.ezuce.com>
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