I think I got this worked out. I had two issues. 1. Inbound Nat on the firewall was correct. Nat settings in Sipx where also correct. However, outbound connections from the sipx server where going out a different external ip address. Thats becuase the NAT I had created was just an inbound static nat. The provider must ignore the SIP message and simply send the message back to the originating ip. Once I sent the outbound nat address to the same as the inbound nat outbound calls worked. 2. But then Inbound calls stopped working. My provider vitality has a different inbound and outbound proxy. I tried every combination of setting the inbound proxy on the main address of the gateway and the outbound proxy under advance but that did not work. So I create two gateways, one for inbound and the other for outbound. Seems to be working now. Not sure if thats the best way to do it but it's working. In testing I did notice a small issue..if a PSTN caller calsl the auto-attendant, then dials extension but hangs up before the person answers, the phone continues to ring until someone answers in then that call is disconnected. I will say this, all in all having a B2BUA in sipx will be great. We deploy a lot of asterisk simply from a cost scenario of not needing a SBC but now with a "fairly" simply SBC in sipx there won't be much need. But now to some more testing. -Matt
>>> On 7/9/2008 at 5:25 PM, in message <[EMAIL PROTECTED]>, "M. Ranganathan" >>> <[EMAIL PROTECTED]> wrote: On Wed, Jul 9, 2008 at 4:21 PM, M. Ranganathan <[EMAIL PROTECTED]> wrote: > Hi, > > > On Wed, Jul 9, 2008 at 3:56 PM, Matt White <[EMAIL PROTECTED]> wrote: >> That was it. Once I sent the profile SipxBridge sprang to life. That link >> you sent was a great help.....too bad I was unable to find by browsing the >> wiki. >> >> Inbound calls seem to be working, a few times I got a "CHANUNAVAIL" but if I >> hang up and call right back it goes through. So i'm not sure if thats a >> provider issue or not. Calls transferring seems to work fine. Hi, I think I can see what is going on here based upon the log you sent me (although next time please send a zip file). You have a misconfiguration. Your address setting in your trunk configuration does not match the domain of the configured ITSP. Please check the Address you set in your siptrunk settings and make sure it matches the proxy domain of the ITSP account ( the proxy domain can be a suffix of the siptrunk address setting if needed ). You may have to look in the Advanced settings to get to the domain setting (there is a change in the works) so the screen may not exactly match what you see on wiki. Ranga >> >> But, I can't get outbound calls working. It's probably in my dial plan. >> But I'm also seeing a 404 denied so I wonder if perhaps the outbound proxy >> is not working. >> >> > > > May I request you to send the trace as an attachment ( zipped ) so I > can look at it through the sipviewer and tell you what the problem > could be. I'll need the proxy trace as well. Please set the log level > to DEBUG for sipxbridge and also set the proxy log level to INFO to > generate the logs. > > Please send the entire directory as a zip file. And yes, further > discussion should be on the dev list. > > Thanks! > > Ranga > >> >> >> Ranga >>> _______________________________________________ >>> sipx-users mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users >>> Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users >>> >> >> >> >> -- >> M. Ranganathan >> >> This email was Anti Virus checked by the Summit Technology Consulting Groups >> Astaro Security Gateway. http://www.astaro.com >> >> > > > > -- > M. Ranganathan > -- M. Ranganathan This email was Anti Virus checked by the Summit Technology Consulting Groups Astaro Security Gateway. http://www.astaro.com
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