On Tue, May 5, 2009 at 4:02 AM, James Holmes <[email protected]> wrote: >> I'm new to sipXecs and I'm having problems getting it to register with >> either of my ITSPs (Vitelity and Les.net). I'm not sure if the problem >> is firewall/NAT related or if I simply have something configured wrong. >> >> When I configuring the Devices->Gateway->Vitelity->SIP Trunk->ITSP >> Account settings I used the Vitelity template for vitelity and the >> les.net template for les.net. I didn't fill any of the advanced settings >> in, just the same username and password as I did when I was using >> trixbox. Also since I'm behind a router with NAT, I went to >> System->Server->[myhostname]->NAT and turned on Use STUN with the STUN >> server set to stun01.sipphone.com, interval 60 and port 5060. I then >> went to Internet calling and selected Enable NAT traversal and Server >> behind NAT. >> >> I'm reasonably sure that the STUN server is working fine because this >> appears regularly in my sipXproxy.log: >> >> StunClient::getPublicIpAddress obtained public IP address 207.161.86.206 >> from server stun01.sipphone.com (which is the correct IP) >> > > Okay I've made some progress on this problem. I'm now able to receive > inbound calls on my LES.NET trunk. As long as I only go to voice mail > they work fine. However this still leaves three problems: > > 1) Vitelity doesn't work. I don't think it's registering. However since > LES.NET is, that suggests that it probably isn't a NAT issue (I may be > wrong here).
There may be something wrong with the template. Please try different settings with the template. It is possible that the ITSP changed settings since the time the template was written. The templates are meant as configuration hints. > > 2) If I pick up the call at my extension (Aastra 57i) instead of letting > the call go to voice mail the call drops after 3 seconds (this doesn't > happen if I let it go to voice mail). Never tested sipxbridge with Aastra 57i. If you send me a sipx-snapshot output I can take a look. > > 3) My outbound dialing doesn't seem to work. The best I can get is "No > route is available to complete this call" message from an operator - the > same operator that I heard when having dialing problems under trixbox. You may want to look at your dial plan configuration. It would be instructive to take a look at the signaling that is leaving sipxbridge and going to the ITSP. Please take a look using sipviewer and in particular take a look at the INVITE being sent out from sipxbridge to the ITSP. 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 _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
