I started with an Audiocodes gateway back in October, it was the one model (FXO+FXS) that sipxecs wouldn't configure and the sipxecs configuration stuff for FXO required it to be treated as a homogenous group of ports. Two things led me to return it:
1) The documentation and manual configuration of the SPA3102 is pretty good compared to Audiocodes (there were numerous occasions when changing what appeared to be a completely unrelated setting resulted in no dialtone on the FXS side, I think they just internally bail if anything is amiss and give you no diagnostics). 2) On a brand new unit they wanted me to buy a service contract to get the current firmware and download the manuals (such as they are) The SPA may be a buggy POS but Audiocodes was at least as frustrating to configure and, as a bonus, it was expensive too. I expect someone using a model supported by sipXecs for configuration would have a better experience. I feel your pain, the SPA sure is a PITA to get going. Happy to help if I can, all those hours spent beating my head on the damn thing might as well go to some good :) -Eric Varsanyi On Feb 18, 2010, at 11:46 AM, [email protected] wrote: > This ebay auction is starting to look tempting :) > http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=350265531020&rvr_id=&crlp=1_263602_263622&UA=%3F*I7&GUID=d74220781250a0e203017dc1fbe784da&itemid=350265531020&ff4=263602_263622 > > Audio Codes MP-114 FXO VOIP Gateway - NEW > US $249.99 > > > On 2/18/2010 11:17 AM, Eric Varsanyi wrote: >> 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 n ot 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/
