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 sti ll 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/
