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/
