I have everything working except what I assume is a dialing rule problem. As soon as I hit send on the Ploycom, I do see the call transferred to the IP of the SPA. If I have dialed a 11 digit number (1 + area code + xxx-xxxx) the call rings immediately. If I dial a 10 digit number (area code + xxx-xxxx) it starts ringing in about 6 seconds. If I dial a 7 digit number, the call doesn't start ringing for 10 seconds. Nothing I have done with the dialing rule seems to change anything. I'm assuming the PSTN Line is the place I need to change this. Interdigit Short Timer defaults to 5 and Interdigit Short Timer: defaults to 10. After reading what they do, I thought that had to be it for sure. http://www.bhatt.id.au/blog/fix-sipura-voip-ata-adapter-interdigit-dialing-delay/ I tried lowering those. It didn't seem to affect anything. I'm assuming that as soon as it shows the IP on the polycom, the call has been transferred to the SPA, so the change I need to make would have to be in the SPA. Any ideas?
On 2/18/2010 11:55 AM, Eric Varsanyi wrote: > 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 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/
