This thing has so many 'sides' I have probably said something incorrect at some point. I'm trying to use it as an outbound gateway only. I'm connecting a POTS line to it and want to be able to make outbound calls from sipx over a POTS line. I believe that all falls under FXO. given the different behavior, I'm sure the hint is what I'm missing. i have put a dial plan everywhere I can find to put one. I can't find any email from you with a working config. if you would resend it, that would be great.
On 2/18/2010 6:50 PM, Eric Varsanyi wrote: > Apologies, I thought you were talking about the FXO side. 2 stage dialing for > FXO is (as I understand it from the "docs") where the first portion of the > dialing comes in via SIP then the user gets a dialtone from the next hop and > dials manually. I didn't pay a lot of attention to that section but I > remember lots of options around delays and waiting for dialtone. > > On the FXS side it makes sense there might be a delay, just like in the > Polycoms (and Pattons) you can give the SPA a hint as to what is a 'complete' > number and the default hint ends with a timeout after N digits. There's a > little script (which I never messed with) in one of the config fields that > defines when to 'send' the call. I thought you were trying to get some > internal SIP device (like a polycom) to dial out on the FXO side and there > was this delay problem there. > > I thought I sent you the config of my "working" 3102 configuration a while > back, the list ate it but I sent it again directly to you. Its just a screen > cap of every config page of my unit. Ask me again if you want it offlist and > I'll send it directly or post it somewhere you can download it from. > > -Eric > > On Feb 18, 2010, at 5:37 PM, [email protected] wrote: > > >> Do you mean to 2 different places to define the dialing plan? If not, I'm >> not sure what 2 stage refers to. >> I didn't see a PDF you sent. Did I miss something? I can't find anything. >> I plugged a handset into the phone port on the SPA, and it behaves the same >> way when I dial form a handset. >> >> On 2/18/2010 4:38 PM, Eric Varsanyi wrote: >> >>> Maybe something related to the 2 stage dialing config? I didn't notice any >>> delays like this using the config I sent in that PDF but I was just >>> thrilled it could make calls at all and might just not have noticed the >>> delay. Maybe plug in a butt-set or a parallel phone and listen for where >>> the delay is to narrow it down (delay seizing line, delay before dialing, >>> delay or slow dialing of digits, ...?). >>> >>> -Eric >>> >>> On Feb 18, 2010, at 4:33 PM, [email protected] wrote: >>> >>> >>> >>>> 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 st ill 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/
