If you can directly use one the configurations from Jim Canfield (and he's got a bunch of them on the wiki) its pretty easy to configure w/o understanding, especially if your'e familiar with any Cisco IOS like configuration paradigm. He's got configurations posted for the straightforward tasks that are easily editable. If you want to do anything at all off the beaten path you kind of have to skim the (excellent) manual and grok how the thing works internally. There are all these internal objects that each serve some purpose in the call chain (in each direction, in and out) and you connect them together like legos. The configuration is pure greek until you understand what the parts you're hooking together are meant to do, once you sort of get what parts you have to work with the thing is pretty straightforward (and a veritable swiss army knife of functionality).
The debugging is excellent, just like IOS you turn on debugging classes and messages of varying detail come to you on the telnet/ssh/serial port session you're logged in on. The more detailed debugging levels are very verbose and its pretty easy to tell what's gone wrong with them turned on. I wrote a couple of pages up on the new wiki about configuring some fancier scenarios (door phone, two different providers all using one 4 port FXO) that I'm using. This page ( http://wiki.sipfoundry.org/display/~evarsanyi/Patton+4114+FXO+5.4+multipurpose+configuration ) has a little diagram I drew up to illustrate the plumbing concepts involved. -Eric Varsanyi On Feb 18, 2010, at 7:18 PM, [email protected] wrote: > Has there ever been an effort to configure sipx so it can configure the > Patton devices? I'm sure they are easy to configure once you have done it > once, but I was told on here the learning curve can be steep. That is why I > chose the audiocodes for my first gateway i had to do. I had it done in 5 > minutes without having to know everything about how it worked. > > On 2/18/2010 6:38 PM, Tony Graziano wrote: >> >> 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/
