Nope. I like to have something that senses disconnects properly. Is factory supported. That is highly configurable. Patton's that ticket for me.
Imagine a text config file where you simple replace one or two pieces of information everytime you add one, and paste the config in. It's a wonderful life. I never spend more than 5 minutes when I get one in from unboxing to deploying. On Thu, Feb 18, 2010 at 6:39 PM, [email protected] < [email protected]> wrote: > 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/ >> >> > > -- ====================== 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.
_______________________________________________ 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/
