It is possible. They have a million options, I know that much. Pressing pound does force the call to dial immediately.
I know. It isn't a great device. For the price, and for what I need it to do, it is worth a try. I have 2 audiocodes MP 114s on the way too, but I know you aren't a big fan of audiocodes either. On 2/18/2010 4:43 PM, Tony Graziano wrote: > Methinks these devices suck rottens eggs. > > In any case, what happens if you end the dialed number with its dial string > termination character (#?)? Does that speed it up? > > When I need a cheap device I cringe, because the time ain't worth the > troubles. > ============================ > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > ----- Original Message ----- > From: [email protected] > <[email protected]> > To: [email protected]<[email protected]> > Cc: [email protected]<[email protected]> > Sent: Thu Feb 18 17:38:33 2010 > Subject: Re: [sipx-users] spa3102 for outbound calls > > 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/ > _______________________________________________ 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/
