Todd,
I appreciate your response, and so quickly too. ;) I have successfully configured x-lite to work with the providers. all of them actually. albeit I cannot test call transfers (maybe I should kick for the full version). I have followed explicitly, over and over again. for a week or more pretty much non stop (changed firewalls, reinstalled, checked dns, etc). the SIP Trunking wiki page. not to mention scouring the internet for information, engaging various people on the list (Thanks Ranga, Tony and others) http://list.sipfoundry.org/archive/sipx-users/msg15966.html I see Voxitas is on the list but callcentric is too. Here is what their support has to say about it. "We have reviewed our logs, and from our side, while we are seeing outgoing calls place under your account; it seems that the calls are failing due to a Network Failure. Are you using your Sipx behind a NAT? We have had trouble with this particular IP PBX, when used behind a NAT however on a public IP it seems to be more reliable." I'll admit that Voxitas is the provider that I have spent the least amount of time with. but believe me that is a decent amount of time. perhaps I am really dense. perhaps I am missing something really obvious. perhaps I cannot figure out how to transfer the settings that work in x-lite into the sipx configuration??? But something is definitely not clicking and after all the time I've invested I'm starting to think I'm crazy. I'd be happy to rebuild the entire test network up from scratch, provide all of the configs, whatever it takes; but there has to be some very low level detail that is screwing with me here??? Thanks, Jonathan Ontra LLC www.ontraonline.com _____ From: Todd Hodgen [mailto:[email protected]] Sent: Thursday, August 06, 2009 4:53 PM To: 'Jonathan Petersen'; [email protected] Subject: RE: [sipx-users] SIP Trunking woes... Jonathan, >From a troubleshooting standpoint, you might want to simplify your testing by simply using a Soft Client and then ensuring you can get things up and running with the soft client first. That will ensure your authentication is working, etc. Baby steps. Once you have verified that any ITSP that you want to test on the sipXecs works on your soft client, then use the "SIP Trunking with sipXecs: Overview and Configuration" to try configuring each one on the sipXecs to the same configuration used on the soft client. http://sipx-wiki.calivia.com/index.php/SipXbridge_Overview_and_Configuration I see Voxitas.com is on the list of providers that have been tested and validated, so I would personally focus on using that provider. There is good information on that provider with their required parameters. Search the user list for emails with voxitas, as I recall a few email exchanges with others that have gotten them working as well. You'll have a "ah hah!" moment when you get it up and running. From: [email protected] [mailto:[email protected]] On Behalf Of Jonathan Petersen Sent: Thursday, August 06, 2009 4:36 PM To: [email protected] Subject: [sipx-users] SIP Trunking woes... Hello list, I have been encountering a lot of problems with SIP trunking and think I need some hand holding. My original local ITSP doesn't support PAI and I was having issues with reinvites (call transfers and hold on incoming calls was broken but worked on outgoing calls?) so I have been looking for other solutions. Currently I have test accounts with callcentric, voxitas, and commpartners configured on sipx. I get inbound calls just fine and can place them on hold, transfer, etc. (which solves the problem with the old provider) but when I go to make outbound calls, they don't even get connected. I believe this is an authentication issue and have requests in for support. but thought I would try here as well to see if anybody is available to walk me through the config and troubleshooting? Thanks, Jonathan Ontra LLC www.ontraonline.com
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