| 00:03.41 | *** join/#asterisk dms (~Adium@c-73-131-162-128.hsd1.sc.comcast.net) |
| 00:07.24 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 00:35.06 | talntid | ~coredump |
| 00:35.07 | infobot | rumour has it, coredump is the guy who maintains the hentges distribution @ http://www.hentges.net |
| 00:36.26 | wyoung | what? |
| 00:36.34 | talntid | was just trying to get info on core dumps |
| 00:36.41 | talntid | after a reboot, my * as doing coredumps |
| 00:36.44 | talntid | lame-o |
| 00:36.50 | wyoung | ah ok |
| 00:36.55 | wyoung | sounds lame |
| 00:36.58 | WIMPy | ~collectdebug |
| 00:36.58 | infobot | collectdebug is probably a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 00:37.51 | talntid | Program received signal SIGILL, Illegal instruction. |
| 00:37.52 | talntid | 0x00000000004d984f in cfmtime_new (who_asked=0x602693 "", filename=0x7fffffff9a00 "/etc/asterisk/asterisk.conf") at config.c:1454 |
| 00:37.52 | talntid | 1454 cfmtime = ast_calloc(1, |
| 00:37.56 | wyoung | hi WIMPy, do you have your hoverboard? |
| 00:37.59 | talntid | apparently, this is what is happening |
| 00:38.15 | WIMPy | wyoung: That was yesterday. |
| 00:38.32 | wyoung | Yeah I didn't get mine yesterday, wondering if you got it today |
| 00:38.44 | WIMPy | That looks evil. Disk contents shreddered? |
| 00:39.03 | talntid | i dunno, shouldn't be... was working fine... |
| 00:39.03 | WIMPy | Nope. Looks like they are delayed. |
| 00:39.11 | talntid | i revooted, and it is doing this |
| 00:39.14 | talntid | super weird |
| 00:40.01 | talntid | in syslog: |
| 00:40.02 | talntid | Oct 22 00:32:14 ip-172-31-30-97 kernel: [ 63.745012] traps: asterisk[1964] trap invalid opcode ip:4d984f sp:7fff7f875720 error:0 in asterisk[400000+295000] |
| 00:41.15 | WIMPy | Well, if you didn't change the CPU something bad happened to at least one important file. |
| 00:43.10 | talntid | i didn't change the cpu |
| 00:43.18 | talntid | and i even did a recompile |
| 00:43.27 | talntid | after the issue presented itself |
| 00:43.32 | WIMPy | thought so. |
| 00:43.36 | talntid | insanely weird |
| 00:43.46 | WIMPy | Then it's some library. |
| 00:45.36 | robmal | htop has a great feature of strace'ing the process live, try it. |
| 00:45.47 | talntid | the process won't even open |
| 00:46.06 | talntid | the moment i try to run it, it says Illegal instruction (core dumped) |
| 00:46.17 | WIMPy | strace |
| 00:46.24 | robmal | Exactly. |
| 00:46.29 | WIMPy | Or just reinstall all libs. |
| 00:46.46 | WIMPy | Or the backup. |
| 00:46.51 | WIMPy | If you have one. |
| 00:47.12 | talntid | http://pastebin.com/KgsyQixQ |
| 00:47.20 | talntid | what libs would i reinstall? |
| 00:47.23 | talntid | i did make uninstall |
| 00:47.26 | talntid | then make install |
| 00:47.48 | talntid | seems to have something to do with realtime clock |
| 00:48.27 | WIMPy | Looks like file-I/O or parsing. |
| 00:48.44 | WIMPy | Do you have a backup? |
| 00:48.56 | talntid | negative, i was just begining this build |
| 00:49.20 | talntid | i can just rebuild it, but what is concerning is this could have happened during a normal production environment |
| 00:49.53 | WIMPy | What FS? What mount options? |
| 00:50.10 | talntid | /dev/xvda1 on / type ext4 (rw,discard) |
| 00:50.48 | WIMPy | Oh, wait. Was that a normal reboot? |
| 00:50.56 | talntid | yes |
| 00:51.00 | talntid | just a normal reboot |
| 00:51.06 | WIMPy | Ouch. |
| 00:51.08 | talntid | working *, then rebooted it. |
| 00:51.16 | talntid | to make sure everything comes back normal |
| 00:51.23 | talntid | and nope, coredump |
| 00:51.29 | WIMPy | Congratulations. You have a component in your system that can't be trusted. |
| 00:51.47 | talntid | scary thing is, i have many * deployments on the same hardware: EC2 |
| 00:52.09 | robmal | How many? |
| 00:53.27 | talntid | mmmm, 6 |
| 00:53.32 | talntid | this would be 7. |
| 00:53.41 | talntid | one of them runs a call center with 70 agents. |
| 00:53.46 | robmal | You should go buy 7 pairs of brown trousers. |
| 01:01.56 | *** join/#asterisk troyt (~troyt@2601:681:4601:cb11:44dd:acff:fe85:9c8e) |
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| 02:08.30 | DivideBy0 | talntid: what instance type do you have for that 70 agent box? |
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| 15:54.59 | *** join/#asterisk infobot (ibot@69-58-76-73.ut.vivintwireless.net) |
| 15:54.59 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.6.0 (2015/10/09), 11.20.0 (2015/10/09), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 15:57.49 | *** join/#asterisk kasanop (~kasanop@ivanpoddubny.com) |
| 16:00.33 | *** join/#asterisk superscrat (asanders@nat/digium/x-rmdwljpqjjusfiic) |
| 16:01.51 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-xravlfxwlhfadaof) |
| 16:05.29 | stefan27 | There's no "FindBugs"-tool for dialplan, something that could search extensions.conf for bad syntax? |
| 16:05.56 | WIMPy | It's called Asterisk :-) |
| 16:08.43 | stefan27 | yeah asterisk complains when the dialplan is executed but it takes ages to test all dialplan |
| 16:10.13 | *** join/#asterisk ruied (~ruied@2001:8a0:774c:1301:56ee:75ff:fe29:4649) |
| 16:10.28 | WIMPy | Yes. |
| 16:11.01 | WIMPy | Well, some syntax errors are shown when loaded, but indeed only some. |
| 16:12.24 | stefan27 | asterisk seems forgiving with some syntax errors as well, like missing ) paranthesies |
| 16:13.23 | WIMPy | Yes. The parser is pretty oppurtunistic. |
| 16:13.35 | stefan27 | i was just curious if anyone has made a static code analyzer script, not just for syntax errors but for common pitfalls |
| 16:14.42 | robmal | What is a common pitfall in a configuration file everyone has different? |
| 16:14.55 | robmal | Except 'you have no extensions.conf' |
| 16:18.26 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 16:20.51 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 16:21.28 | igcewieling | Generally Asterisk expects you to provide a valid extensions.conf |
| 16:22.57 | igcewieling | you could try conf2ael -- it might catch some of the errors you are looking for. |
| 16:24.22 | stefan27 | thanks |
| 16:24.44 | stefan27 | robmal i dont know what's common, perhaps calling apps in wrong order, playback before answer |
| 16:25.46 | rmudgett | playback has a valid use before answer. Also playback will answer itself if the channel is not answered and it has not been explicitly told to not answer. |
| 16:26.23 | stefan27 | yeah I just took that out of the blue |
| 16:26.30 | rmudgett | Each application parses its own parameters so a generic dialplan syntax checker cannot exist. |
| 16:27.54 | davidbowlby | [TK]D-Fender - I have opened my firewall and can talk them through doing it on their side, but I thought there was a way to do it that doesn't require firewall rules as long as their outbound rules aren't restrictive (normally the case) |
| 16:29.59 | davidbowlby | Are there phones with built in VPN clients that get around the issue? |
| 16:30.01 | stefan27 | I made one mistake of calling an app with app(${VAR1},${VAR2}) where VAR1 could contain commas |
| 16:30.17 | davidbowlby | stefan27, time to switch to pipes ;) |
| 16:30.43 | stefan27 | but it worked when one passed them as "${VAR1}" |
| 16:32.13 | nbjoerg | davidbowlby: ALG with SIP support on the firewall normally handles that? |
| 16:32.19 | *** join/#asterisk kritzikratzi (~kritzikra@62-46-86-225.adsl.highway.telekom.at) |
| 16:32.24 | nbjoerg | (of course, doesn't help if you use TLS) |
| 16:32.30 | [TK]D-Fender | "firewall" != proper settings |
| 16:32.42 | davidbowlby | nbjoerg, client side or server side? |
| 16:32.48 | stefan27 | I'm sure there are some patterns for mistakes, like Set(uniquevarname=boo) followed by a Dial |
| 16:32.52 | nbjoerg | davidbowlby: between client and server |
| 16:33.12 | nbjoerg | basically, the classic FTP problem of having a second channel again... |
| 16:33.14 | davidbowlby | nbjoerg, but the client has their own gateway, think home user |
| 16:33.20 | stefan27 | to a local channel, that need not be a mistake but I often forgot underscores when i want underscored scope |
| 16:33.44 | nbjoerg | davidbowlby: yes and most home routers do have SIP ALGs configured |
| 16:34.00 | stefan27 | but I get your points, if a syntax-checker _had_ made a lot of sense, someone would have made it :) |
| 16:34.54 | davidbowlby | njboerg, if they don't, would STUN work around that? |
| 16:35.20 | stefan27 | didn't commas replace pipes in asterisk 12 or somewhere, davidbowlby? |
| 16:35.43 | davidbowlby | stefan27, no no, nm me |
| 16:35.58 | davidbowlby | stefan27, it was joke, hence the ;) |
| 16:37.17 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:37.30 | stefan27 | ah |
| 16:38.36 | [TK]D-Fender | stefan27didn't commas replace pipes in asterisk 12 or somewhere, davidbowlby? 1.4 <- |
| 16:38.49 | nbjoerg | davidbowlby: IIRC, it's one of the possible options of making things more firewall friendly. symmetric RTP helps a bit as well |
| 16:38.51 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 16:38.51 | *** mode/#asterisk [+o file] by ChanServ |
| 16:38.59 | *** join/#asterisk jonno11 (~Jon@cpc1-walt12-2-0-cust582.13-2.cable.virginm.net) |
| 16:39.15 | nbjoerg | davidbowlby: also disable media handover |
| 16:39.47 | stefan27 | Oh, i see [TK]D-Fender, I only ever tried asterisk1.2 12 and 13 |
| 16:39.48 | davidbowlby | nbjoerg, I get the problem and how to resolve it via firewall rules, I'm just looking for a way to not require at home folks to touch their routers |
| 16:40.28 | [TK]D-Fender | stefan27: Try not to assume we wasted 10 years before getting those changes :) |
| 16:40.38 | davidbowlby | nbjoerg, I've seen vonage work without touching the firewall and just wondered if they use a different approach. |
| 16:40.55 | [TK]D-Fender | davidbowlby: You virtually NEVER have to touch a remote user's router |
| 16:41.13 | [TK]D-Fender | davidbowlby: Unless it is horribly buggy or hostile |
| 16:41.22 | [TK]D-Fender | davidbowlby: and butting in (ALG, etc) |
| 16:41.31 | [TK]D-Fender | Fix your peer settings |
| 16:41.38 | davidbowlby | [TK]D-Fender, in what way |
| 16:41.49 | [TK]D-Fender | Tell it that it is behind NAT |
| 16:41.59 | [TK]D-Fender | send your KEEP ALIVE so taht you can continue comms with it. |
| 16:42.04 | [TK]D-Fender | Prevent reinvites |
| 16:42.04 | davidbowlby | ah crap, one sec, maybe I forgot lol |
| 16:42.32 | davidbowlby | I did that with my interconnects, but may have missed it on the endpoint |
| 16:42.42 | davidbowlby | bangs head on table |
| 16:43.24 | [TK]D-Fender | "Am I supposed to trust the media IP's in the invites"? , "Do I need a keep-alive in order to contact them later or will NAT close up behind me?", "NAT'd devices cannot be allowed to reinvite. Did I screw up and allow something so silly as this?" |
| 16:43.58 | [TK]D-Fender | that is just the PEER side |
| 16:44.14 | [TK]D-Fender | Now if your SERVER is behind NAT ... that is a pile MORE settings |
| 16:44.29 | davidbowlby | [TK]D-Fender that I have working fine, server is good |
| 16:44.41 | [TK]D-Fender | Sometimes "working" is not "good" |
| 16:44.54 | [TK]D-Fender | And is just waiting for the proepr circumstance to bite you in the ass |
| 16:44.59 | davidbowlby | [TK]D-Fender lol I mean things work fine if the firewall on the client side is set to route to the phone |
| 16:45.03 | [TK]D-Fender | and "good" doesn't answer that yes/no question... |
| 16:45.21 | davidbowlby | [TK]D-Fender, this is what I'm preparing for, don't want to have to modify the client firewall for every home |
| 16:45.28 | [TK]D-Fender | cliens should almost never need forwarding |
| 16:45.31 | [TK]D-Fender | And you should ne be doing |
| 16:45.47 | davidbowlby | [TK]D-Fender, phone with firewall rule can register, dial, audio, everything works |
| 16:46.05 | [TK]D-Fender | remove the forwarding. and configure your peer right |
| 16:46.12 | davidbowlby | [TK]D-Fender, phone behind a wifi router without firewall/port forwarding configured doesn't register |
| 16:46.18 | [TK]D-Fender | You are over exposing it and complicating the setup |
| 16:49.46 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
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| 17:01.26 | davidbowlby | [TK]D-Fender >:( so, umm, lol peer settings were it |
| 17:01.53 | davidbowlby | [TK]D-Fender some days I could screw up a free lunch |
| 17:03.14 | *** join/#asterisk Jesterboxboy (~Thunderbi@124.128.220.106) |
| 17:04.08 | Jesterboxboy | If i have something like |
| 17:04.08 | Jesterboxboy | exten => _434343.,1,Set(DIALED=${EXTEN:6}) |
| 17:04.08 | Jesterboxboy | in my dialplan and someone is calling 434343666, asterisk should set DIALED to 666 ? |
| 17:04.48 | Jesterboxboy | iam asking because asterisk is not matchin the extension 434343666 here in my dialplan |
| 17:06.35 | davidbowlby | jesterboxyboy, is it hitting the right context? |
| 17:08.03 | Jesterboxboy | yes |
| 17:08.51 | Jesterboxboy | i see it hitting the correct context, if i change it to |
| 17:08.51 | Jesterboxboy | exten => 434343,1,Set(DIALED=${EXTEN:6}) |
| 17:08.51 | *** join/#asterisk dms (~Adium@162.207.112.26) |
| 17:08.54 | Jesterboxboy | its working |
| 17:09.12 | Jesterboxboy | but i just cant figure out why its not matching with the above? the syntax shoudl be right or? |
| 17:09.25 | Jesterboxboy | _434343. should catch all numbers starting with 434343? |
| 17:09.38 | [TK]D-Fender | so far... |
| 17:09.43 | [TK]D-Fender | Show us the actual call and actual dialpan |
| 17:09.48 | [TK]D-Fender | ~pb |
| 17:09.49 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:09.50 | [TK]D-Fender | ^^^ |
| 17:10.37 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 17:10.44 | Jesterboxboy | not quite comfortable with pasting the number, just wanted to ask if anyone is seeing a syntax error here? |
| 17:11.35 | nbjoerg | . or .! ? |
| 17:14.39 | WIMPy | . or ! - .! doesn't make sense. |
| 17:16.48 | igcewieling | Jesterboxboy: _434343. should catch all numbers starting with 434343 followed by ONE OR MORE chars, but not match just 434343 |
| 17:17.05 | Jesterboxboy | ahh! |
| 17:17.07 | Jesterboxboy | thanks |
| 17:17.24 | Jesterboxboy | how would i write this? |
| 17:17.32 | WIMPy | You could argue that "starting with" implies that :-) |
| 17:17.35 | igcewieling | Jesterboxboy: common n00b mistake |
| 17:17.43 | Jesterboxboy | yep i bow my head in shame |
| 17:17.57 | igcewieling | Jesterboxboy: you can't. you need one for 434343 and one for "_434343." |
| 17:18.07 | igcewieling | yes it is ugly, no there is no way aroun dit. |
| 17:18.11 | WIMPy | Arg! |
| 17:18.11 | Jesterboxboy | okay thanks |
| 17:18.20 | WIMPy | You use _434343! . |
| 17:18.36 | igcewieling | WIMPy: I did not know that worked on SIP. |
| 17:18.51 | WIMPy | It has nothing to do with channeltype. |
| 17:19.11 | igcewieling | Jesterboxboy: you can use his method or mine. |
| 17:19.21 | igcewieling | use whichever works. |
| 17:19.28 | WIMPy | But, yes, sip overlap and overlapping extensions is not possible. |
| 17:20.18 | Jesterboxboy | why would need !. and not just ! |
| 17:22.05 | igcewieling | Jesterboxboy: have you read extensions.conf.sample? |
| 17:22.50 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 17:22.59 | Jesterboxboy | no, iam just going through https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 17:23.05 | Jesterboxboy | but i will do that now |
| 17:24.42 | WIMPy | There are two differences between . and ! - one is that ! will match nothing. |
| 17:24.55 | [TK]D-Fender | igcewielingyes it is ugly, no there is no way aroun dit. <- in many cases, there is |
| 17:25.34 | [TK]D-Fender | and you never mix "." with "!" |
| 17:25.41 | Jesterboxboy | @WIMPyokay but only if taken on its own? not if 23423! ? |
| 17:25.43 | [TK]D-Fender | they are both terminators |
| 17:26.25 | [TK]D-Fender | Depends if you ar dialing like an IVR, or if that's a context a call initially lands on with a complete number or not |
| 17:26.39 | WIMPy | _23423! will match 23423, so that would be nothing. |
| 17:26.54 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:27.18 | WIMPy | Not from the Asterisk side, no. |
| 17:27.53 | rmudgett | Another important difference between "." and "!" is that "!" will NOT wait for more digits to arrive. It matches immediately. |
| 17:27.55 | igcewieling | Yes, it is SFW: http://www.fix6.net/wp-content/uploads/2011/03/mine-is-bigger-than-yours-front.png |
| 17:28.47 | WIMPy | igcewieling: Thats nice. |
| 17:28.59 | WIMPy | rmudgett: Unless there are other extensions that couls also match. |
| 17:29.14 | igcewieling | [TK]D-Fender: I have vague memories of pain and misery associated with !, but I last used it so many years ago...... |
| 17:30.13 | [TK]D-Fender | Works fine for direct dialed complete numbers, not "as-you-go" |
| 17:31.03 | WIMPy | yes, but that's the idea, isn't it? Otherwise you'd use . . |
| 17:31.30 | WIMPy | But if you use sip you are extremely unlikely to notice. |
| 17:32.19 | igcewieling | Jesterboxboy: Im interesting in knowing how ! works out for you, let us know. |
| 17:32.29 | Jesterboxboy | it worked out good for me |
| 17:32.45 | Jesterboxboy | catches the main number plus all extensions |
| 17:33.39 | Jesterboxboy | i have one other question: can i use a logical and in asterisk dialplan |
| 17:34.11 | Jesterboxboy | something like: |
| 17:34.11 | Jesterboxboy | GotoIf($[${LEN(${DIALED})} >1 && $LEN(${DIALED} <5]?label |
| 17:34.51 | WIMPy | If you want that, I'd make extensions for all allowed lengths instead. |
| 17:35.18 | [TK]D-Fender | Jesterboxboy: Yes there is a logic "AND" |
| 17:35.28 | [TK]D-Fender | read the WIKI page on Asterisk Expressions |
| 17:35.39 | WIMPy | I.e. _1234X, _1234XX, _1234XXX and _1234XXXX . |
| 17:36.01 | WIMPy | No need to accept the calls first. |
| 17:37.25 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 17:42.26 | [TK]D-Fender | Technically he'd just use 2 patterns |
| 17:42.33 | [TK]D-Fender | one with ".", one without |
| 17:42.39 | [TK]D-Fender | if he wanted it IVR-able |
| 17:42.56 | WIMPy | Not if he wants a maximum length. |
| 17:43.22 | [TK]D-Fender | true |
| 17:45.01 | WIMPy | It might however make sense to ignore additional digits for DDI if you have any sort of IVR. |
| 17:48.50 | [TK]D-Fender | Well we're missing a little bit of usage info and dialing circumstance so.... |
| 17:49.19 | *** join/#asterisk areski (~areski@80.174.128.34.dyn.user.ono.com) |
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| 17:53.40 | rl1 | i got this nortel mg1010 and dunno what to do with it |
| 17:53.46 | rl1 | probably sell it on ebay |
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| 18:30.38 | jameswf | https://usercontent.irccloud-cdn.com/file/M0pmYIbe/ |
| 18:34.54 | kfife | Question: In Asterisk 12/13, if I have res_speech.so loaded, do I need anything other than grammar files to do ASR? |
| 18:35.03 | kfife | I'm trying to do some simple digit recognition |
| 18:36.16 | kfife | IOW: years ago you had to explicitly connect to a speech engine like Lumenvox. I'm trying to understand if there has been a concept change |
| 18:39.14 | file | Asterisk isn't a speech engine, you have to have something else |
| 18:41.08 | kfife | the digium wiki seems to refer to a default speech engine. What's your recommendation for a lightweight application like digit recognition? |
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| 19:06.00 | igcewieling | kfife: where does it say that? |
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| 20:55.50 | jonm | Can anyone help me locate a standalone appliance or PCIE card to do 4 POTS lines -> my asterisk box? Ive been trying to browse through Sangoma stuff but I'm not that familiar with analog / DAHDI |
| 20:56.36 | robmal | You're looking for fxo/fxs |
| 20:57.51 | jonm | Is that like a Sangoma A104 ? |
| 20:59.14 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 20:59.17 | jonm | Or is that more like the A200 series with 4 ports. |
| 21:01.58 | [TK]D-Fender | yes |
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| 21:02.53 | jonm | So both would work ? =p |
| 21:03.15 | jonm | seems like the a104 might be the better option depending on price |
| 21:04.19 | [TK]D-Fender | a014 = digital |
| 21:04.23 | [TK]D-Fender | Completely different |
| 21:04.56 | jonm | youre right, i just re-read it. the a200 with 4 ports seems to be the one. |
| 21:09.05 | jonm | And just to make sure - i want FXO since its not providing dialtone and phones arent plugging into the ports, it is the "end unit" |
| 21:15.39 | igcewieling | Adtran should have a product to do what you want, but they are not cheap. |
| 21:20.38 | Phil-Work | jonm, I've been using one of these: http://www.ebay.co.uk/itm/Quad-Span-Analog-Card-For-asterisk-card-PCI-FXO-Card-FXS-Card-FXO-Board-x400p-/161792957139?hash=item25ab9c9ad3:g:wsEAAOSwu4BV0VU- |
| 21:21.12 | Phil-Work | bit slow to get from China so I'd always have a spare on hand for a production system |
| 21:21.28 | Phil-Work | but it's been doing the job for 2 years now |
| 21:21.42 | WIMPy | Before you buy such a card, you want to make sure that you don't have any other choice. |
| 21:22.02 | jonm | Wow thats ludicrously cheap =p. This is being explicitly used for fire/911 though, and I know Sangoma has a stellar reputation around here. |
| 21:22.06 | igcewieling | The trouble with a card like that is you get no support. From anyone. |
| 21:22.25 | WIMPy | And if you still think that's your only choice, make sure it really really is your only choice. |
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| 21:24.23 | Phil-Work | I'd think that with any card, regardless of price, if it were being used irregularly for E-911 I'd have it perform an automated test call every hour or so |
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| 21:44.09 | malcolmd | i like to think digium's own A4B cards are pretty handy for PCI-Express and FXO. i'm certainly biased, but it does continue putting coins into asterisk development. :D |
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