So after 2 hours on the phone we figured it out. Basically because the two SIP trunks are using the same domain, when we use one or the other trunk we get a bad authentication error, as if one trunk is trying to overwrite the other trunks username and password. As soon as we setup another trunk under a different domain (ITSP) they both work as they should. There is a ticket being generated from Nortel to the design team to fix this issue J
Thanks Derek Bartolo Technical Support Comsource Technologies 3585 Laird Drive Unit 13 Mississauga Ontario [email protected] <mailto:[email protected]> 905-608-1400 ext 240 From: [email protected] [mailto:[email protected]] On Behalf Of Tony Graziano Sent: Wednesday, November 04, 2009 12:40 PM To: Burden, Mike Cc: [email protected] Subject: Re: [sipx-users] FW: Multiple gateways? You need to find out from your ITSP what they expect to see from you in an outgoing call (number format), ITSP templates are not one size fits all. You also need to make sure you know what they are sending you for an inbound call (number format). Since your ITSP is not on the template list, guessing will never work well. Can you receive any incoming calls? I mean a trace by this: http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Si pviewer On Wed, Nov 4, 2009 at 12:31 PM, Burden, Mike <[email protected]> wrote: I am using the same dialing plan that was working correctly with the previous ITSP - I simply added the new Gateway to it and removed the old Gateway. I have tried dialing 7 digits, PSTN Prefix + 7 digits, 10 digits, PSTN Prefix + 10 digits, 11 digits, and PSTN Prefix + 11 digits, all with the same result. I'll see what I can do about getting a call trace. I assume you're talking about a Wireshark trace? From: Tony Graziano [mailto:[email protected]] Sent: Wednesday, November 04, 2009 12:13 PM To: Burden, Mike Cc: [email protected] Subject: Re: [sipx-users] FW: Multiple gateways? It sounds as if they are authenticating you by IP address. If so, can you receive calls? A call-trace would be handy too. With bandwidth.com, we use the default company (main telephone) number as the outgoing caller id by placing the number as the username (no password). What is the format of your dialed number? Does the gateway automatically add a "+1" to all calls, if so, make sure you dial "just" the 10 digits so the +1 gets added automatically. On Wed, Nov 4, 2009 at 12:08 PM, Burden, Mike <[email protected]> wrote: Local Exchange Carriers of Michigan. From: Tony Graziano [mailto:[email protected]] Sent: Wednesday, November 04, 2009 12:04 PM To: Burden, Mike Cc: [email protected] Subject: Re: [sipx-users] Multiple gateways? Who is the ITSP? On Wed, Nov 4, 2009 at 12:01 PM, Burden, Mike <[email protected]> wrote: The new ITSP is not assigning a username/password, and is simply authenticating us by our source IP address. When I set up the Gateway for the new ITSP, I simply left the username and password blank. Is that the correct thing to do? From: [email protected] [mailto: [email protected]] On Behalf Of Burden, Mike Sent: Wednesday, November 04, 2009 12:00 PM To: [email protected] Subject: Re: [sipx-users] Multiple gateways? As a test, I removed the Gateway for the SIP Trunk to the first ITSP, leaving only the Gateway to the new ITSP. I'm still getting a fast busy, so apparently I have a problem with the Gateway configuration for the new ITSP. Could that prevent the sipXecs from placing an outbound call correctly, even though there was another, working Gateway specified ahead of it in the Dial Plan? From: [email protected] [mailto: [email protected]] On Behalf Of Burden, Mike Sent: Wednesday, November 04, 2009 11:47 AM To: [email protected] Subject: [sipx-users] Multiple gateways? Good morning, OK, I'm brand new to both IP telephony and sipXecs, so if I've done something dumb, go easy on me. :) I have sipXecs working correctly with a SIP trunk to our ITSP. This Gateway uses the Internal SBC (sipXbridge-1) as the route. I want to add a SIP trunk to a second ITSP. I went to Devices -> Gateways -> Add New Gateway -> SIP Trunk and filled in the information for the new ITSP, and set up the Dial Plan so that the original Gateway was first, followed by the Gateway to the new ITSP. The new Gateway also uses shipXbridge-1 as the route. After restarting services, outbound calls do not work at all (fast busy.) I removed the Gateway for the new ITSP from the Dial Plan and restarted services, but it still didn't work (still fast busy.) I removed the new Gateway and restarted services again, and everything worked again. Where am I going wrong? Shouldn't the system use the Gateways in the order I specified in the Dial Plan? Mike Burden Lynk Systems, Inc e-mail: [email protected] Phone: 616-532-4985 _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/
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