Gotcha. I do find it interesting that it behave the exact same way when I dialed. I will put a splitter on the port tomorrow and attach a butt set to listen in. I got your PDF. I will take a look at it.
On 2/18/2010 7:31 PM, Eric Varsanyi wrote: > When you connected a phone to the 'telephone' jack on the SPA that was the > FXS side, that side cares about dial plans (its the thing that gives dialtone > and 'sends' the call on the SIP side when it thinks it has the whole number). > For debugging I was suggesting you attach a phone in parallel to the FXO jack > (or use a high impedance butt-set) and listen in to see what the timing of > the linksys vs the pots line vs you hitting send on the polycom sounds like. > The idea is to try to narrow down where the delay is happening. > > I'll send my pdf config in the next email. Sorry for the confusion. > > -Eric Varsanyi > > > On Feb 18, 2010, at 7:13 PM, [email protected] wrote: > > >> 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 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/
