For debugging if you set it up to send syslog messages and turn the level all the way up it sometimes produces semi-useful output. You don't have to have a syslog server set up to catch it if you can run tcpdump or socat.
If you can capture traffic to/from the device with tcpdump that's probably the next step if the syslog stuff doesn't pay off (it kind of sounds like either its ignoring you or sipxproxy isn't really sending the invite where you hope its going). -Eric Varsanyi On Feb 18, 2010, at 11:09 AM, [email protected] wrote: > I have no doubt it is a PEBKAC, I just don't where to look. I changed it to > 5061 (I now see that setting in the PSTN Line tab on the spa3102). The logs > look about the same to me. I don't see anything that even tells me it is > making it to the spa3102. > > On 2/18/2010 10:54 AM, Eric Varsanyi wrote: >> When I set mine up late last year the only issue I had making outbound calls >> (that wasn't PEBKAC) was the thing didn't think there was a line attached >> and returned something like 'resource not avaiable' to the invite. I had to >> change the line voltage threshold down in the international settings box to >> fix this. >> >> Ah, in the log I see you're using 5060, the FXO side by default is on 5061 >> (the FXS is on 5060). LIkely that's your issue. >> >> -Eric >> >> On Feb 18, 2010, at 10:42 AM, [email protected] wrote: >> >> >>> I'm trying to configure a spa3102 for outbound calls only. I know people >>> pull their hair out over these devices, but I wanted to give it a shot. My >>> only gateways I've worked with so far are sipxbridge and an audiocodes >>> configred from within sipx, so I haven't really done too much manual FXO >>> configuration. >>> I think I may be missing something on the sipx end, because I don't think >>> the call is ever making it to the spa3102. This is a new setup and has no >>> other gateways. I added the spa3102 as an unmanaged gateway. I enabled all >>> the dialing plans and added the gateway. I'm using a polycom 550, Sipx >>> 4.0.4, bootrom 4.2.1, firmware 3.1.3C split. I would show a siptrace, but >>> the merged file doesn't really have anything in it. The sipx server is at >>> 10.81.1.5. The spa3102 is at 10.81.1.6. I tried setting the gateway in sipx >>> to UDP manually (that is what the spa3102 defaults to) and specifying port >>> 5060, but that didn't seem to change anything. There are only 2 logs >>> created, so I attached those. Is there something simple I'm missing? I read >>> through this, >>> http://sipx-wiki.calivia.com/index.php/HowTo_configure_sipX_to_work_with_FXO/FXS_SIP_Gateways >>> but I don't see anything that sticks out at me. The only thig I thought I >>> might need to do is something in authrules.xml, but I'm still not sure since the text around it refers to FXS and this is FXO. I sort of guess there has to be some some sort of authorization for the spa3102 to know the sipx call can be sent outbound, but I don't know where to do this. Sorry if I'm missing something obvious here. I think the fact that I got an audiocodes 8 port working inbound and outbound with no questions (and clearly not much knowledge on the subject) is a testament to how well sipx is able to configure it! >>> >>> Thanks, >>> Matthew >>> <sipregistrar.log><sipXproxy.log>_______________________________________________ >>> 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/ >>> >> > > <sipregistrar.log><sipXproxy.log> _______________________________________________ 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/
