I have a lot of clients who ask me to come in and explain those to folks, auditors too. Add to that my kids say my favorite word is "No.". I shoot so straight people put on a a Bad Company album when I walk in the room. Move over Lord Vader.
On Thu, Feb 18, 2010 at 6:14 PM, Eric Varsanyi <[email protected]> wrote: > Yeah, I wish Patton had a 1 port FXO at twice the price of an SPA3102. A > Patton 1FXO+1FXS with failover relay would remove any desire to suffer at > the hands of Linksys. > > What really kills me about Cisco/Linksys is the hardware isn't bad and the > firmware would actually be OK (if a little unintuitive in the UI dept) if it > wasn't riddled with bugs. They still sell these things (and other ones in > the same line) but they've completely abandoned even basic ongoing > maintenance. At least Amazon took the ones I had bought back as 'Defective > by design' (for a full refund). > > -Eric > > On Feb 18, 2010, at 4:54 PM, Nathaniel Watkins wrote: > > > One thing I'll give Tony - he doesn't beat around the bush :) > > > > -----Original Message----- > > From: [email protected] [mailto: > [email protected]] On Behalf Of Tony Graziano > > Sent: Thursday, February 18, 2010 5:44 PM > > To: [email protected]; [email protected] > > Cc: [email protected] > > Subject: Re: [sipx-users] spa3102 for outbound calls > > > > 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-dia > >> ling-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=d74220781250a0e203017dc1fbe7 > >>>> 84da&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/ > > > > This message and any files transmitted with it are intended only for the > individual(s) or entity named. If you are not the intended individual(s) or > entity named you are hereby notified that any disclosure, copying, > distribution or reliance upon its contents is strictly prohibited. If you > have received this in error, please notify the sender, delete the original, > and destroy all copies. Email transmissions cannot be guaranteed to be > secure or error-free as information could be intercepted, corrupted, lost, > destroyed, arrive late or incomplete, or contain viruses. Garrett County > Government therefore does not accept any liability for any errors or > omissions in the contents of this message, which arise as a result of email > transmission. > > > > > > Garrett County Government, > > 203 South Fourth Street, Courthouse, Oakland, Maryland 21550 > www.garrettcounty.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/ > -- ====================== 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/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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