| 00:00.16 | mmmitch | sorry, i must be being extremely dense |
| 00:01.05 | mmmitch | you mean http://www.shiffman.net/itp/asterisk/Client.java ? |
| 00:07.33 | *** join/#asterisk YoMomma (~YoMomma@cpe-98-149-120-227.socal.res.rr.com) |
| 00:17.13 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 00:19.42 | Dar1us | mmyes :) |
| 00:19.47 | Dar1us | ah well too late |
| 00:20.27 | Dar1us | pity since it's obvious what's wrong :-/ |
| 00:29.31 | *** join/#asterisk Sophira (~sophie@neo.theblob.org) |
| 00:29.38 | Sophira | Hiya. |
| 00:30.08 | Sophira | Is there any way in a dialplan to check if *any* key exists in a given DB family? |
| 00:31.19 | ectospasm | Sophira: DBEXISTS()? |
| 00:32.54 | Sophira | ectospasm: I thought that just does it for a single key. I could be wrong, though. |
| 00:33.07 | Sophira | gives it a test. |
| 00:36.56 | Sophira | Yeah, I get an error. |
| 00:36.57 | Sophira | func_db.c:204 function_db_exists: DB_EXISTS requires an argument, DB(<family>/<key>) |
| 00:37.14 | Sophira | This is with: |
| 00:37.14 | Sophira | <PROTECTED> |
| 00:37.14 | Sophira | <PROTECTED> |
| 00:37.33 | Corydon76-work | Sophira: what version? |
| 00:38.03 | Corydon76-work | Try it with a single trailing slash character |
| 00:38.05 | Sophira | Oh, hey. DB_EXISTS(testtest/) works. |
| 00:38.12 | Sophira | Heh. |
| 00:38.35 | Sophira | I'm using 10.7.0. |
| 00:38.37 | Sophira | But yeah. |
| 00:38.41 | Sophira | The slash seems to work. |
| 00:39.09 | *** join/#asterisk YoMomma (~YoMomma@cpe-98-149-120-227.socal.res.rr.com) |
| 00:39.23 | Sophira | Thanks Corydon :) |
| 00:40.08 | Corydon76-work | yw |
| 00:40.26 | ectospasm | Corydon76-work is the man |
| 00:40.42 | Corydon76-work | I used to be the man. Now I'm the zombie |
| 00:40.48 | Sophira | ...never mind, I did it wrong, actually. |
| 00:40.51 | zombu2 | now now now |
| 00:40.57 | Sophira | Turns out it doesn't work after all :/ |
| 00:41.13 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:41.14 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:41.29 | Sophira | <PROTECTED> |
| 00:41.29 | Sophira | <PROTECTED> |
| 00:41.46 | Sophira | So yeah, sorry to get your hopes up :/ |
| 00:41.46 | Corydon76-work | Asterisk 10 is weird when it comes to astdb, because it's actually implemented with sqlite |
| 00:41.47 | [TK]D-Fender | family & key aren't the same in both |
| 00:41.53 | [TK]D-Fender | And you should be entering both... |
| 00:42.10 | Sophira | [TK]D-Fender: But that's the thing. I want to test if *any* key exists in this family. |
| 00:42.18 | [TK]D-Fender | Sophira: No way for this |
| 00:42.30 | Sophira | Aw. Okay. |
| 00:42.37 | [TK]D-Fender | Sophira: Not directly within the dialplan anyway |
| 00:43.00 | ectospasm | you could write an AGI that greps the underlying sqlite db |
| 00:43.05 | [TK]D-Fender | Sophira: Perhaps you could reconsider what youre doing in a way that doesn't have thinking about having to do this. |
| 00:43.10 | Corydon76-work | Sophira: what's your use case anyway? |
| 00:43.17 | Sophira | [TK]D-Fender: I'm probably doing this the wrong way anyway. |
| 00:43.23 | Sophira | Yeah, okay, let me explain. |
| 00:43.29 | [TK]D-Fender | Sophira: 11 steps to go ;) |
| 00:44.12 | Corydon76-work | I've used astdb for some really funky stuff, but the long and short of it is, you probably ought to be using a real database. |
| 00:44.46 | Sophira | I want to know if somebody else is already calling this particular DID. But I want to do that after I've already done some after stuff, if possible. |
| 00:45.38 | Corydon76-work | You might want to consider using something like the extension state for that |
| 00:45.52 | Sophira | The way I was thinking of doing it such that it doesn't have too much of a problem with race conditions was Set()ing a variable with the name being a filtered caller ID of the caller (which I've already verified is being passed), then remove it on hangup. |
| 00:45.57 | Corydon76-work | astdb is rather stale information |
| 00:46.08 | Sophira | Okay. |
| 00:47.18 | Sophira | I don't know much about extension states, but I know it involves hints. The trouble is, I would basically like to only check to see if somebody has made it past a certain part of the dialplan, I guess. |
| 00:47.40 | Sophira | Tell you what. |
| 00:47.43 | Sophira | I know how to do this. |
| 00:47.48 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-64-66.pppoe.mtu-net.ru) |
| 00:47.51 | Corydon76-work | You could use LOCK() and TRYLOCK() for that, too |
| 00:48.08 | Corydon76-work | You're essentially trying to put a mutex on the DID, right? |
| 00:48.14 | Sophira | Oh hey, that's a thing. |
| 00:48.15 | Sophira | Yes. |
| 00:48.32 | Corydon76-work | And locks automatically clear when the channel dies |
| 00:48.43 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:48.44 | Sophira | That's great, actually. |
| 00:48.47 | Sophira | Thank you <3 |
| 00:49.06 | Corydon76-work | yw. Doubly so, since I wrote LOCK/TRYLOCK |
| 00:49.42 | Sophira | TRYLOCK returns instantly, I'm guessing? |
| 00:49.52 | Corydon76-work | Correct |
| 00:49.56 | Sophira | Awesome. |
| 00:50.08 | Corydon76-work | with a value that specifies whether the lock succeeded or not |
| 00:52.15 | Sophira | Okay. So what I'll probably do to make sure it's as fast as possible, but also that no race conditions occur, is to TryLock first, and if there's no lock, Lock() it. If that fails due to a race condition, then I can go somewhere based on that. |
| 00:53.00 | Corydon76-work | Nope, TRYLOCK is identical to LOCK, except that it does not hang out and try to get the lock |
| 00:53.13 | Sophira | Oh, okay. |
| 00:53.24 | Corydon76-work | If the name is unlocked, TRYLOCK and LOCK behave identically |
| 00:53.28 | Sophira | My mistake, I thought it queried the status. Okay, that sounds good too. |
| 00:53.53 | Corydon76-work | Can't, because as soon as you query a status, it's possible that information is already stale |
| 00:54.06 | Sophira | Yeah, hence what I was saying about race conditions. |
| 00:54.16 | Sophira | But glad to know that's not necessary :D |
| 00:55.42 | Sophira | BTW, how come it's a function and not an application? |
| 00:55.54 | General_Z0D | anyone that can point me to a current/modern tutorial for setting up asterisk realtime with mysql? something that shows or has the sql tables as well |
| 00:58.14 | Corydon76-work | General_Z0D: ls -l contrib/realtime/mysql/ |
| 00:58.28 | General_Z0D | what does that mean? |
| 00:58.40 | Corydon76-work | Type that command from within the source directory |
| 01:00.08 | General_Z0D | will those create the main db as well or do i just manuall make one called asterisk. The last tiem i played with this stuff was asterisk version 1.4 |
| 01:00.18 | General_Z0D | so things im sure have changed a bit |
| 01:00.26 | Corydon76-work | Realtime hasn't changed that much |
| 01:01.15 | General_Z0D | all the tutorials out there are either improper sql syntax or missing the sql files so im glad i have those now perhaps i can get a2billing going thanks Corydon76-work |
| 01:01.27 | Corydon76-work | yw |
| 01:05.49 | General_Z0D | actually do you know of a good billing platform that is decent and open source or is a2billing pretty much it? |
| 01:06.59 | Corydon76-work | I don't keep up. Certainly it's the most well-known |
| 01:07.05 | General_Z0D | and is realtime configuration covered in the asterisk definitive guide? |
| 01:07.15 | Corydon76-work | But billing platforms should be moving towards CEL and not CDR. |
| 01:07.16 | General_Z0D | i just want to be sure i follow something current |
| 01:07.20 | [TK]D-Fender | Did you look at the table of contents? |
| 01:07.40 | General_Z0D | not yet right now I am just asking questions while i gather the files together |
| 01:07.53 | [TK]D-Fender | You answer is no, mine is yes. |
| 01:08.55 | [TK]D-Fender | Corydon76-work: Yeah CEL is going to be better for "real" billing in a transfer scenario, but CDR will do for most chump CC-type businesses... which we know is the regrettable majority |
| 01:09.02 | General_Z0D | yeah chapter 16 |
| 01:09.58 | Sophira | Corydon76-work: Quick question - is the mutex obtained by LOCK/TRYLOCK released before or after the "h" hangup extension is executed? |
| 01:10.05 | General_Z0D | well i am setting up a small voip provider for disadvantaged people, and phone based tech support along the same lines i figured something like a2billing would be fine for my needs |
| 01:10.23 | Corydon76-work | Sophira: After, when the channel is destroyed |
| 01:10.33 | Sophira | Okay, awesome, thanks. :) |
| 01:10.56 | Corydon76-work | If you want it removed earlier, you need to use UNLOCK() |
| 01:11.29 | Corydon76-work | If a DID goes to voicemail, I'd suggest calling UNLOCK() before invoking Voicemail() |
| 01:25.39 | *** join/#asterisk Nemus (~Nemus@c-76-23-30-50.hsd1.ut.comcast.net) |
| 01:32.54 | *** join/#asterisk Nemus (~Nemus@c-76-23-30-50.hsd1.ut.comcast.net) |
| 01:57.37 | metabsd | hi, wich version of asterisk + freepbx you suggest for production environnement >? |
| 01:57.40 | metabsd | Thank you! |
| 01:59.32 | WIMPy | I'm not sure anyone in here would do so at all. Try in #freepbx. |
| 01:59.49 | zombu2 | anyone know how to force a payload size for video calls? |
| 02:05.14 | metabsd | ok asterisk version for production ? 10.X 1.8.X ? |
| 02:05.37 | WIMPy | They're both ok for me. |
| 02:05.37 | metabsd | LTS = 1.8.X |
| 02:05.44 | WIMPy | If you want to do conferencing, use 10. |
| 02:06.04 | metabsd | i use conferencing for paging |
| 02:06.16 | metabsd | meet_me |
| 02:07.19 | zombu2 | you guys any clue about video payload sizes? |
| 02:07.19 | metabsd | i install 1.8.11 cert4 last month but now Digium release another version. |
| 02:07.27 | WIMPy | ConfBridge, the better MeetMe. |
| 02:07.52 | metabsd | WIMPy ok but i read about ConfBridge and it's feature less for the moment... |
| 02:08.08 | WIMPy | That was in 1.8, not in 10. |
| 02:08.17 | WIMPy | Hence my original comment. |
| 02:08.28 | metabsd | in 10.X it's better ? because in 1.8.X i dont have hangup feature in confbridge .. |
| 02:08.54 | WIMPy | In 10 you can put custom menus in there. |
| 02:09.34 | metabsd | i will install a lab with 10.x but another thing about this version ... the next LTS is 11.X en january 2013 you think its a good idea to use 10.X in production environnement ? |
| 02:10.05 | WIMPy | You can already use 11 if you want LTS :-) |
| 02:10.26 | metabsd | the stable release for 2013 .. not now |
| 02:10.45 | metabsd | i want stability :-) |
| 02:11.15 | WIMPy | All 3 should be fine. |
| 02:11.28 | zombu2 | sigh* |
| 02:11.29 | metabsd | ok thank you! |
| 02:36.08 | *** join/#asterisk luckman212 (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1) |
| 02:39.08 | *** join/#asterisk luckman212 (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1) |
| 02:50.28 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:50.33 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:05.03 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 03:05.04 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:20.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 03:24.33 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 04:01.46 | *** join/#asterisk YoMomma (~YoMomma@cpe-98-149-120-227.socal.res.rr.com) |
| 04:18.04 | *** join/#asterisk nomad411 (~nomad411@76-10-128-207.dsl.teksavvy.com) |
| 04:21.25 | nomad411 | Hi everyone. I'm a web consultant/UNIX guy. One client asked me to install Asterisk on their Web server (A virtual dedicated server at an ISP) . Does that sound like something that might be possible? I'd normally not put such varied services on a same server, but what do I know? Wouldn't the Asterisk server need access to a landline? |
| 04:22.49 | nomad411 | By the way, my reply has been "I'd rather not see multiple services liek this on the same machine", and that it,s not my expertise. I'd rather NOT get into this on the webserver. There must be systems that can be bougth and setup at their restaurant to serve their phoe services? |
| 04:22.52 | nomad411 | phone |
| 04:22.54 | ChannelZ | not unless they just used it for VoIP |
| 04:23.52 | ChannelZ | and yes there are Asterisk "appliances" you can buy |
| 04:23.53 | nomad411 | It sounds liek they want to use it for all their phone needs |
| 04:24.06 | ChannelZ | see digium.com |
| 04:24.27 | nomad411 | ok.. I'm there now. :) |
| 04:25.39 | nomad411 | I think I need to do way more research :) |
| 04:26.40 | ChannelZ | Well unless Asterisk is something you want to take a few months to learn and support, I'd just refer them to some other telecom consultant |
| 04:27.42 | ChannelZ | I don't have any personal experience with the Switchvox systems but even with a GUI it'd take some time to get familiar with I imagine |
| 04:28.01 | nomad411 | ChannelZ: It's my feeling as well. He,s just hoping I'll take it on for free, for the immense benefits this experience will bring me |
| 04:28.15 | nomad411 | but it's not what I do :) I think I have someone to hook them up with |
| 04:28.39 | nomad411 | Last thing I need is to be woken in the middle of the night because a phone doesn't worj |
| 04:28.51 | ChannelZ | but if they have more than a handful of phones to support, it'd more than likely be better off on a separate server depending on what the load of that virtual one happens to be |
| 04:29.53 | nomad411 | I think they want a nice wide switchboard with varied voicemails and redirects to various phones |
| 04:30.22 | *** join/#asterisk dmz (~dmz@unaffiliated/dmz) |
| 04:31.00 | nomad411 | Thanks for the confirmation ;) |
| 04:31.12 | nomad411 | Off to email people who do this already |
| 04:31.33 | ChannelZ | Asterisk isn't real hungry unless you're running a lot of phones, but latency is everything. Has to have good internet connectivity |
| 04:31.55 | ChannelZ | good luck |
| 04:32.43 | nomad411 | If there are ways to have that ISP server connect to the phone system, without a physical line, then maybe it would woek on their server |
| 04:33.24 | nomad411 | Cheers! Good night/day |
| 04:40.10 | *** join/#asterisk coppice (~chatzilla@m121-202-10-9.smartone.com) |
| 04:46.35 | ChannelZ | too late, but "yes" |
| 04:49.00 | *** join/#asterisk YoMomma (~YoMomma@cpe-98-149-120-227.socal.res.rr.com) |
| 04:51.10 | *** join/#asterisk mintos (mvaliyav@nat/redhat/session) |
| 05:27.28 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 05:30.49 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:06.14 | *** join/#asterisk kikohnl (~keith@udp278187uds.hawaiiantel.net) |
| 06:07.39 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 06:24.25 | *** join/#asterisk wierdo (d4193396@gateway/web/freenode/session) |
| 06:28.03 | *** join/#asterisk irroot (~gregory@2002:c5ac:2ac7:3:50cc:90dc:836d:286a) |
| 06:28.27 | *** part/#asterisk irroot (~gregory@2002:c5ac:2ac7:3:50cc:90dc:836d:286a) |
| 06:40.11 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 06:40.19 | v0lZy | morning |
| 06:40.47 | v0lZy | i just noticed something and i have no idea whats going on... but my verbosity is set to 9 |
| 06:41.06 | v0lZy | and when i reload the dialplan i just get == Parsing '/etc/asterisk/extensions.conf': == Found |
| 06:41.11 | v0lZy | after that, when i make a call |
| 06:41.15 | v0lZy | I get a single line |
| 06:41.27 | v0lZy | == Using SIP RTP CoS mark 5 |
| 06:41.44 | v0lZy | and thats it |
| 06:41.44 | v0lZy | i dont see any dialplan running at all |
| 06:41.45 | v0lZy | I dont see any of the noops i put it etc, nothing |
| 06:41.49 | v0lZy | just that and a busy tone. |
| 06:42.15 | ChannelZ | sighs |
| 06:43.20 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 06:43.33 | ChannelZ | sounds like your extensions are hosed and didn't actually get loaded |
| 06:43.41 | ChannelZ | for crying out loud... |
| 06:44.44 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 06:45.26 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 06:46.09 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-emcvvkulqmlppehd) |
| 06:46.13 | *** join/#asterisk wierdo (d4193396@gateway/web/freenode/ip.212.25.51.150) |
| 06:46.33 | ChannelZ | v0lZy: sounds like your extensions are hosed and didn't actually get loaded if all you saw was 'parsing extensions.conf' and didn't see every extension get added with verbose on |
| 06:46.53 | v0lZy | hosed? |
| 06:47.17 | ChannelZ | 'fucked up' |
| 06:47.39 | v0lZy | hm |
| 06:47.50 | v0lZy | I started the file with a comment |
| 06:47.54 | ChannelZ | syntax error early in the file, or some foreign character |
| 06:47.56 | v0lZy | a really long comment |
| 06:48.06 | v0lZy | But i thought i can have comments above [globals] |
| 06:48.11 | v0lZy | apparently they cant be too long |
| 06:48.12 | ChannelZ | you can |
| 06:48.38 | ChannelZ | sure your editor didn't line-wrap it onto an actual second line? |
| 06:49.28 | ChannelZ | There very well could be a buffer limit |
| 06:49.34 | v0lZy | I broke the comments down myself |
| 06:49.37 | v0lZy | ;like this |
| 06:49.40 | v0lZy | ;and this |
| 06:49.42 | v0lZy | ;and so on |
| 06:49.48 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 06:49.57 | ChannelZ | well that should be fine but something else is probably going on |
| 06:50.20 | v0lZy | hm |
| 06:50.22 | v0lZy | funny |
| 06:50.23 | v0lZy | now it work |
| 06:50.25 | v0lZy | s |
| 06:50.59 | v0lZy | hm |
| 06:51.00 | v0lZy | ok |
| 06:51.04 | v0lZy | apparently there's two things |
| 06:51.15 | v0lZy | u cant have too many lines of comments above [globals] |
| 06:51.38 | v0lZy | and, asterisk doesnt reload the dialplan if its fucked up in that way but keeps the previous one going until restarted |
| 06:57.20 | kaldemar | you can have as many lines of comments as you like. you just did something wrong. |
| 07:01.07 | ChannelZ | still suspects a magical mystery character |
| 07:01.32 | ChannelZ | (like a CR/LF you aren't seeing or something) |
| 07:05.11 | v0lZy | i've been using mcedit |
| 07:05.23 | v0lZy | and i've backspaced every line to the last character in it |
| 07:05.57 | v0lZy | im not sure of that kaldemar. Reason is, if i move the lines below [globals], it works ok |
| 07:06.05 | v0lZy | if i keep them above globals, no gam |
| 07:06.07 | v0lZy | e |
| 07:06.16 | ChannelZ | Any profanity? Asterisk is very sensitive. |
| 07:06.24 | v0lZy | profanity? |
| 07:06.32 | v0lZy | in my comments? no. |
| 07:06.46 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 07:07.03 | kaldemar | v0lZy: i am. |
| 07:07.45 | kaldemar | v0lZy: the parser is more tolerant for uncommented stuff under globals. |
| 07:08.21 | v0lZy | http://bpaste.net/show/Uk9LD2KwOBQYNmdQnrNk/ |
| 07:08.26 | v0lZy | here's what i have |
| 07:08.30 | v0lZy | cant see anything wrong with it |
| 07:09.53 | kaldemar | v0lZy: you have : instead of ; on line 172. |
| 07:10.26 | v0lZy | huh |
| 07:10.38 | v0lZy | missed that one |
| 07:10.40 | v0lZy | thanks kaldemar |
| 07:10.46 | kaldemar | this stuff is why you should start by pastebinning your stuff instead of having a guessing competition. |
| 07:11.51 | kaldemar | letting asterisk output warnings and notices to CLI won't hurt either. see logger.conf. |
| 07:12.42 | v0lZy | works now |
| 07:12.47 | v0lZy | thanks. |
| 07:12.53 | v0lZy | logger.conf, ill check it out |
| 07:13.28 | v0lZy | by the way, u can see what im doing |
| 07:13.31 | v0lZy | i documented the stuff there |
| 07:17.15 | v0lZy | Any thoughts on it? still dont like it? |
| 07:17.22 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:18.47 | kaldemar | new names on stuff that are already named and artificial division and namings. |
| 07:21.47 | v0lZy | yeah i know, im not trying to rename things or anything, I'm just writing it down the way its easy to understand for me, or others like me which might be completely new to anything related to how telephony works |
| 07:22.31 | v0lZy | I'd be happy to get the proper 'speak' for the stuff i re-named. |
| 07:25.03 | kaldemar | when a call reaches dialplan, there is no dialstring, but an incoming extension. and all incoming requests are just as external from asterisk's point of view, whether they come from phones in your LAN or a provider outside your network. and contexts in dialplan are called contexts, not applications. applications are what NoOp and Dial are for example. top-down: dialplan-context-extension-priority-application-function |
| 07:25.39 | kaldemar | dialstring is something that is given to app Dial. |
| 07:27.37 | v0lZy | I found it confusing to refer to it that way since it overlaps with asterisk speak. |
| 07:28.37 | v0lZy | I mean, going on what my trouble is when consulting the documentation etc, a lot of stuff is not 'naturally' explained as to what it is or what it does. The docs are more geared to people who've been in this business before and know this stuff more |
| 07:29.34 | v0lZy | its like hitting somone with DID DAHDI, DUNDI, etc... stuff that he can't guess as to what it would mean without some prior knowledge |
| 07:30.19 | v0lZy | the division to internal/external is more from the perspective of the phone user |
| 07:30.29 | v0lZy | like 'i have to dial 0 to get out' |
| 07:30.54 | kaldemar | eh, for example DID, DAHDI, and DUNDi are all explained in the book. |
| 07:31.11 | v0lZy | don thave the book :| |
| 07:31.22 | v0lZy | something i've been meaning to rectify |
| 07:31.38 | kaldemar | 'i have to dial 0 to get out' is just a choice that someone may make. it is not needed. |
| 07:31.44 | kaldemar | ~book |
| 07:31.44 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 07:31.57 | kaldemar | it's readable on-line. |
| 07:32.06 | v0lZy | oh? |
| 07:32.32 | v0lZy | thanks! |
| 07:33.04 | v0lZy | Well yeah, i know its not needed |
| 07:33.07 | v0lZy | but in the company |
| 07:33.14 | v0lZy | you usually have something like that |
| 07:34.02 | v0lZy | internal telephony network, and then external.... and theres this whole division of what you want the callers on the inside to be able to do and what u want the callers from outside to be able to do, which of the people u want to allow only internal calls, which of the people can call whatever they want etc |
| 07:34.17 | v0lZy | like my boss doesnt want people in the warehouse having access to external lines |
| 07:34.26 | v0lZy | thinking they're gonna call some hotline or something |
| 07:34.43 | v0lZy | but he wants them to have emergency numbers like 911 etc |
| 07:35.11 | v0lZy | on the other hand, he wants his phone to be redirected to our front desk if he doesnt answer it in 20 seconds |
| 07:35.25 | v0lZy | but only if the call is from someone outside our company |
| 07:35.29 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 07:35.51 | v0lZy | for people calling from within the company's network, he doesnt want such redirection |
| 07:36.14 | v0lZy | also, we have voicemail for leaving eachother messages, but no voicemail for users calling in |
| 07:36.25 | v0lZy | Except for outside working hours |
| 07:36.48 | kaldemar | v0lZy: people having access to external lines have nothing at all to do with using 0 to dial out. |
| 07:36.59 | v0lZy | and it needs to be optional on a per user basis because people in the sales department dont want to be obliged to call back customers that are phoning the wrong line for support etc |
| 07:37.19 | kaldemar | v0lZy: that's why you have context and their proper use. that is why i have told you that your approach causes problems. |
| 07:37.51 | v0lZy | kaldemar: i know. its not to restrict access, its to separate internal numbers and external numbers when dialing in my case. first number is choice of provider. |
| 07:38.47 | v0lZy | 0 for dialing out with our provider... 1 for using our sister company in Croatia, and so on |
| 07:39.14 | v0lZy | 9 for our backup line |
| 07:39.27 | *** join/#asterisk UQlev (~chatzilla@46.251.117.25) |
| 07:40.16 | v0lZy | kaldemar: yeah but i have internal users who need access both to internal and external desitnations |
| 07:40.23 | v0lZy | and then some who only should have internal |
| 07:40.44 | v0lZy | I can put them in the same context, but then i have to define for each and every one, what they can and cant call. |
| 07:40.55 | v0lZy | i can still have 2 context |
| 07:41.05 | v0lZy | and use them like groups |
| 07:41.10 | kaldemar | your approach just makes that harder for you now. |
| 07:41.28 | v0lZy | how so? |
| 07:41.33 | v0lZy | I mean.. |
| 07:41.41 | v0lZy | its just a mattter of dialplan entry context |
| 07:41.51 | v0lZy | thats set on the phone. |
| 07:41.59 | v0lZy | ../account. |
| 07:42.33 | v0lZy | I can still do like [PEER16] and just copy stuff from the [DialPatterns] and not include the stuff for external calls. |
| 07:42.50 | v0lZy | and i can remove those numbers if i dont want them receiving external calls from the [IncomingPatterns] |
| 07:44.06 | v0lZy | I started with a gui |
| 07:44.09 | v0lZy | AskoziaPBX |
| 07:44.21 | v0lZy | to copy paste their extensions.conf ... this is the mess i started with: http://bpaste.net/show/B0Ow212Xuw5nr5IPn70U/ |
| 07:44.40 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85ed.bcn.adamo.es) |
| 07:46.01 | v0lZy | i think my dialplan is more clear if nothing else |
| 07:48.35 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:48.59 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:50.14 | kaldemar | anything looks good compared to that. |
| 07:58.51 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 07:59.14 | v0lZy | heh, yeah :D |
| 08:00.32 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:3411:64a5:97f:bfa9) |
| 08:07.05 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 08:19.47 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 08:20.26 | phix | hey, is there a way to differentiate between the user not picking up and issues dialing out through a particular trunk/.' |
| 08:20.29 | phix | ? |
| 08:21.08 | phix | I am having issues dialing out (probably havnt paid bill) and DIAL_STATUS is comming up as CONGESTED, it does that when the user doesn't pick up either |
| 08:22.18 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 08:22.56 | *** join/#asterisk jsjc (~Adium@206.Red-2-136-102.dynamicIP.rima-tde.net) |
| 08:25.04 | wdoekes | phix: check ${HANGUPCAUSE} instead |
| 08:25.20 | wdoekes | see include/asterisk/causes.h for the number/name mapping |
| 08:25.38 | kaldemar | phix: a user not picking up within the timeout set in the Dial app would result in NOANSWER in DIALSTATUS. |
| 08:29.04 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 08:37.02 | phix | wdoekes: This particular asterisk setup is version 1.4, is HANGUPCAUSE availlable in that version/ |
| 08:37.29 | phix | kaldemar: ok, so what would CONGESTED be? the user was on another call? |
| 08:38.26 | kaldemar | see hangup cause for that situation. |
| 08:39.23 | phix | I basically want to try another trunk is the first one fails (except if the issue was the the user being called was busy / engaged, the user hangs up, any other fault / error signal that doesn't mean the trunk failed) |
| 08:40.24 | phix | so if I don't get a response from the trunk, it isn't available or I have no monies left in that account I want it to move to the next trunk |
| 08:42.55 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 08:43.08 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 08:43.32 | v0lZy | doesnt dial have that by default though? |
| 08:43.36 | v0lZy | failover provider |
| 08:46.19 | kaldemar | no. |
| 08:52.01 | v0lZy | phix: i think my askozia has something like that |
| 08:52.08 | v0lZy | its probably cryptic but i can check out the dialplan and paste |
| 08:53.17 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:55.02 | v0lZy | hm |
| 08:55.08 | v0lZy | never mind, i didnt enable it, its not in the config |
| 09:00.56 | *** join/#asterisk racho (2e287bcc@gateway/web/freenode/ip.46.40.123.204) |
| 09:01.39 | *** join/#asterisk Vince-0 (c4d7bcf4@gateway/web/freenode/ip.196.215.188.244) |
| 09:02.08 | racho | hello, can somebody please explain why can't I capture asterisk vars in my python script? I'm using agilib which is part of pyst i think. |
| 09:04.27 | kaldemar | ~ask |
| 09:04.27 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 09:04.38 | *** join/#asterisk longst (~longst@46-22-127-214.bredband.alvsjo.qonet.se) |
| 09:05.10 | longst | if it possible use getVar and SetVar from Asterisk CLI ? |
| 09:08.07 | kaldemar | longst: no. |
| 09:08.10 | phix | v0lZy: I have a dial plan for it, it just seems to be timing out but the call isn't making it, it may be a firewall issue I think:\ |
| 09:08.44 | phix | If I ring my phone number from 2 different VoIP trunks nothing happens |
| 09:09.50 | phix | :/, I am getting alot of these too --> [Jul 22 03:28:48] NOTICE[5910] chan_sip.c: Call from '' to extension '999990011972592231975' rejected because extension not found. |
| 09:10.21 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 09:10.28 | phix | damn script romanians |
| 09:10.40 | phix | romanian script kiddie |
| 09:10.42 | phix | even |
| 09:13.25 | wdoekes | phix: trying if it works takes as much time as asking |
| 09:13.42 | wdoekes | and less of my time |
| 09:15.59 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 09:26.33 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 09:35.13 | *** join/#asterisk Goldwing (~Goldwing@84.245.47.128) |
| 09:37.03 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 09:41.30 | phix | wdoekes: ah yes I was trying it out |
| 09:53.04 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 09:53.28 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:55.47 | [sr] | hi WIMPy |
| 10:04.33 | *** join/#asterisk UQlev (~chatzilla@46.251.117.25) |
| 10:04.35 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-rhwbvxwxsuepwsun) |
| 10:12.27 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 10:18.42 | *** join/#asterisk pyther (~pyther@unaffiliated/pyther) |
| 10:20.31 | engrxyz | els |
| 10:42.33 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-qrommxqsxyikppbf) |
| 11:03.57 | *** join/#asterisk longst (~longst@46-22-127-214.bredband.alvsjo.qonet.se) |
| 11:09.00 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 11:15.10 | WIMPy | Why do days always start too early? |
| 11:19.49 | bulkorok | just wrong timezone... |
| 11:23.16 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 11:25.02 | *** join/#asterisk mirela666 (~mirko@212.200.146.253) |
| 11:25.21 | Roelt | you're always in the wrong timezone |
| 11:25.41 | WIMPy | Probably. |
| 11:26.01 | Roelt | unless you're on a holiday, then the days are too short |
| 11:26.32 | WIMPy | Both days and nights are too short. Always. |
| 11:32.18 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 11:32.24 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 11:35.47 | *** join/#asterisk danfromuk (~IceChat77@2.30.230.189) |
| 11:35.49 | *** join/#asterisk b3nt_pin (~quassel@stjhnf0148w-142134201057.dhcp-dynamic.FibreOp.nl.bellaliant.net) |
| 11:41.43 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:50.53 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 11:52.17 | [sr] | WIMPy: i'm with you |
| 11:59.12 | WIMPy | Cought in the same timezone? ;-) |
| 11:59.29 | [sr] | ya... 1hr less but it's almost the same |
| 11:59.30 | [sr] | :p |
| 12:04.47 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-xxdywjxifkqcpciy) |
| 12:04.48 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:05.26 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-xxdywjxifkqcpciy) |
| 12:05.47 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-xxdywjxifkqcpciy) |
| 12:05.47 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:07.27 | *** join/#asterisk wonderworld (~ww@dsdf-4db53358.pool.mediaWays.net) |
| 12:07.48 | carrar | *Y*A*W*N* |
| 12:16.32 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:17.37 | *** join/#asterisk kessius (~kessius@189.4.61.222) |
| 12:23.32 | mirela666 | does anyone know what is the best (or only) way to do a dial on a sip/user, Playback file to that sip/user and secretly listen to both Playback and SIP/users channel? :) |
| 12:25.39 | WIMPy | Live? |
| 12:27.24 | mirela666 | yes :) |
| 12:27.55 | kaldemar | originates and ChanSpy. |
| 12:29.31 | mirela666 | I tried with Originate but it creates in and out channels and I get Playback on wrong one i guess, |
| 12:29.41 | mirela666 | thx, I'll try it out some more |
| 12:31.31 | kaldemar | originate in such way that the peer gets dialed first, then upon answer connect it to an extension that originates another call that gets connected to ChanSpy and then proceeds to the playback. |
| 12:32.15 | [TK]D-Fender | 2 x chanspy <- |
| 12:32.47 | [TK]D-Fender | Originate the playback one with a 1-2s delay in playback, then fall through to chanspy on a direct exten. |
| 12:33.48 | mirela666 | problem is to get all of that to start at the same time, when peer answers lol |
| 12:34.36 | [TK]D-Fender | You don't DIAL them. You CHANSPY them. There is no delay as there is no "answering" required on their end |
| 12:34.57 | mirela666 | aha |
| 12:51.19 | *** join/#asterisk longst (~longst@46-22-127-214.bredband.alvsjo.qonet.se) |
| 12:54.30 | *** join/#asterisk bchia (~Adium@nat/digium/x-xwjcryvevccwibsm) |
| 12:57.25 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 13:00.43 | *** join/#asterisk serafie (~erin@nat/digium/x-hhpdkoetsrnecxdk) |
| 13:01.07 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:01.07 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:01.50 | *** join/#asterisk rolandow (~roland@92.68.81.83) |
| 13:07.01 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 13:10.10 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 13:10.23 | Faustov | hi, any recommended SIP trunk providers for Canada? |
| 13:13.51 | WIMPy | ~itsplist-ca |
| 13:13.51 | infobot | rumour has it, itsplist-ca is Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca |
| 13:20.45 | mirela666 | http://pastebin.com/Q5ca6fFF |
| 13:20.45 | mirela666 | This is what i got so far |
| 13:21.53 | Faustov | WIMPy: thanks, if that's recommended I'll look into it (the list of providers is huge otherwise) |
| 13:29.21 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:35.29 | Katty | morning |
| 13:35.31 | mirela666 | ok got the right ChanSpy sintax now :) |
| 13:36.22 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 13:41.04 | *** join/#asterisk oej_ (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 13:46.35 | carrar | Morning Katty |
| 13:46.52 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:46.55 | Katty | naps on carrar |
| 13:47.33 | carrar | gets squished |
| 13:52.07 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 13:57.20 | WIMPy | Katty: What's gone wrong with your Asterisk? Does it work today? |
| 13:58.32 | sruffell | smirks |
| 13:58.39 | Katty | too sleepy for asterisk tday |
| 13:58.45 | Katty | WIMPy: how'rechu? |
| 13:58.56 | Katty | and yes it's fine, as always |
| 13:59.03 | WIMPy | Poor Asterisk. |
| 13:59.22 | WIMPy | finally gets some breakfast... |
| 14:06.21 | *** join/#asterisk Tarso (~Cookie_Mo@189.61.47.149) |
| 14:07.20 | *** part/#asterisk Tarso (~Cookie_Mo@189.61.47.149) |
| 14:09.08 | *** join/#asterisk Tarso (~Cookie_Mo@189.61.47.149) |
| 14:11.47 | *** join/#asterisk YoMomma (~YoMomma@cpe-98-149-120-227.socal.res.rr.com) |
| 14:13.20 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 14:13.52 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:13.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:16.49 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 14:16.52 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 14:23.33 | *** part/#asterisk mirela666 (~mirko@212.200.146.253) |
| 14:27.34 | *** join/#asterisk mmmitch (928d0150@gateway/web/freenode/ip.146.141.1.80) |
| 14:28.18 | mmmitch | can somebody run through the reasons again why "write() failed: Resource temporarily unavailable" might be happening using AGI? |
| 14:28.39 | mmmitch | in file.c:1296 waitstream_core |
| 14:28.58 | WIMPy | Because noone is listening? |
| 14:29.22 | WIMPy | But someone was disappointed last time you left, commenting that the issue was obvious. |
| 14:30.23 | mmmitch | oh man |
| 14:31.04 | Goldwing | Q: just wondering if it's possible, i have 2 locations that are connected together thru a bridged openvpn making it one big network. |
| 14:31.33 | Goldwing | on location 1 there is a asterisk server, location 2 connects thru the bridge to the serrver on location one |
| 14:31.34 | WIMPy | That seems possible. :-) |
| 14:32.05 | WIMPy | mmmitch: I cannot comment on your Java stuff, but my first guess is that it doesn't process the audio it receives. |
| 14:32.23 | Goldwing | now, is it possible to have another asterisk server on location 2 that has the clients of location 2 and the servers connect to eachother |
| 14:32.53 | WIMPy | Goldwing: Sure |
| 14:32.55 | Goldwing | on location on we have a SIP trunk for incoming and dailout, i want that same trunk on location 2 |
| 14:33.25 | Goldwing | and that the phones from location 1 can dail to location 2 just if they are on the same server |
| 14:33.34 | WIMPy | Your ITSP will obviousely only send the calls to one location. |
| 14:33.38 | Goldwing | and the other way also offcource |
| 14:34.06 | Goldwing | ok |
| 14:34.30 | Goldwing | so dail-in goed thru location 1, but both locations get a seperate trunk for dailing out |
| 14:34.59 | jeffspeff | Goldwing, is the internet so slow that you have to put a box at each location? |
| 14:35.07 | WIMPy | Unles your account is IP restricted you should be able to call out simultaneousely. |
| 14:35.26 | Goldwing | no, it's more like a failover idea |
| 14:35.28 | Goldwing | both locations have 100mbit fiber |
| 14:36.33 | Goldwing | the main idea is, that sould the openvpn connection be disconnected, or should one location lose internet that the other location can work like nothing is wrong |
| 14:37.14 | Goldwing | i allready did that with the DHCP, DNS and a custom made program |
| 14:37.15 | jeffspeff | i would get a trunk for each location and bridge the boxes through the vpn via iax trunk |
| 14:37.27 | WIMPy | Might make sense to get a second account for the 2nd location. So the calls are already routed byt the ITSP to the correct location. |
| 14:38.00 | WIMPy | But even with only one account registered by both boxes would help. |
| 14:38.36 | WIMPy | Calls would hit any location at random normally and if one goes down, the other will work. |
| 14:38.55 | Goldwing | well, you said, that "send the calls to one location." so if i configure the incoming SIP on both servers, and one disconnects, the other will take over... right? |
| 14:39.04 | Goldwing | yea |
| 14:39.15 | WIMPy | Just if both stay up, but lose connection between each other it will develop "random features". |
| 14:39.36 | WIMPy | The next time it registers, yes. |
| 14:39.49 | Goldwing | and if i make the same dailing rules on both locations, sending the incoming trunk to the same phones, it *would* work.. |
| 14:39.51 | Goldwing | oh.. ok |
| 14:39.52 | Goldwing | gmm |
| 14:39.53 | Goldwing | hmm |
| 14:40.19 | Goldwing | *scrathing head* |
| 14:40.36 | Katty | noms homemade bread |
| 14:40.43 | beek | is jealous |
| 14:40.48 | WIMPy | Unless your ITSP does multiple registration per account. In that case both boxes would get the calls simultaneousely. |
| 14:40.53 | Katty | shares with beek |
| 14:41.02 | Katty | beek: it's amish white bread |
| 14:41.07 | beek | noms homemade bread |
| 14:41.09 | Katty | beek: made it saturday morning |
| 14:41.10 | mmmitch | also struggling with " ast_waitfordigit_full: The FD we were waiting for has something waiting. Waitfordigit returning numeric 1". Possibly same issue |
| 14:41.14 | beek | thanks Katty! |
| 14:41.20 | Katty | beek: do you want the recipe? |
| 14:41.33 | beek | Ummmm, no. I'll just share yours, thanks. |
| 14:41.39 | Goldwing | yea, but that would confuze the system i guess, because 2 incming calls would register on the phones for each call coming in |
| 14:41.46 | WIMPy | mmmitch: Please explain. |
| 14:42.02 | Goldwing | hmm |
| 14:42.24 | WIMPy | Goldwing: Yes. That would only work correctly with a proxy, not a B2BUA like Asterisk. |
| 14:43.44 | Goldwing | back to the drawing board... |
| 14:43.50 | *** join/#asterisk stegbth (~stegbth@2001:6f8:9fa:0:221:70ff:fea9:b5e2) |
| 14:43.57 | stegbth | hello everybody |
| 14:44.16 | stegbth | how can i define the calling extension? |
| 14:44.44 | mmmitch | WIMPy: when I send out "WAIT FOR DIGIT" to asterisk from java, I get back "ast_waitfordigit_full: The FD we were waiting for has something waiting. Waitfordigit returning numeric 1" |
| 14:44.49 | stegbth | i have a script, which sends an eMails and afterwards calls a mobilephone |
| 14:44.55 | WIMPy | stegbth: That question doesn't make sense. Please rephrase/explain. |
| 14:44.57 | stegbth | exten => _X.,n,Dial(Local/${EXTEN}@from-internal/n,${RTCFU},${DIAL_OPTIONS}) |
| 14:45.28 | stegbth | ah , sorry from beginning. |
| 14:45.31 | WIMPy | mmmitch: And what's wrong with that? |
| 14:45.59 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:46.10 | WIMPy | stegbth: Set CALLERID(num) |
| 14:46.10 | stegbth | i a have a trixbox, there i have an extension, which calls out to a mobilephone |
| 14:46.19 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:46.19 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:46.24 | mmmitch | I'm not actually able to parse out the DTMF digits |
| 14:46.24 | WIMPy | ~trixbox |
| 14:46.24 | infobot | Trixbox is unable to be supported here. It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support. Try joining #trixbox and asking your questions there. |
| 14:46.29 | mmmitch | in AGI |
| 14:46.53 | stegbth | ok, (num) is the whished extension? |
| 14:47.22 | WIMPy | stegbth: No literal. You assign the number to it. |
| 14:47.30 | *** join/#asterisk dms (~dms@nat/digium/x-iraljzwclogzxpxv) |
| 14:47.31 | WIMPy | stegbth: Set(CALLERID(num)=08154711) |
| 14:47.41 | stegbth | ah, ok |
| 15:15.21 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.37) |
| 15:15.46 | lorsungcu | yo |
| 15:18.07 | ChannelZ | oy |
| 15:19.16 | *** join/#asterisk bchia (~Adium@nat/digium/x-yikxwyvumbqzyqhi) |
| 15:20.25 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 15:36.38 | *** join/#asterisk roe (~roe@unaffiliated/roe) |
| 15:37.16 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 15:48.49 | roe | Does anyone have a non-partisan opinion of Digium sip phones? |
| 15:52.45 | *** part/#asterisk longst (~longst@46-22-127-214.bredband.alvsjo.qonet.se) |
| 15:54.06 | WIMPy | roe: Maybe not the ideal location for that question. |
| 15:54.28 | roe | WIMPy, I know, but I can't really find any reviews of them online |
| 15:54.40 | roe | and I figured the most likely users of them owuld be asterisk users |
| 15:55.40 | WIMPy | Sure, but if you don't trust them... |
| 15:55.50 | WIMPy | BTW: I wouldn't, either. |
| 15:56.28 | roe | they *look* good |
| 15:57.09 | WIMPy | Nothing that uses SIP can be good. |
| 15:57.40 | roe | that's an odd opinion |
| 15:58.40 | WIMPy | SIP wasn't made for telephony and that's still pretty obvious. |
| 15:59.00 | bchia | roe - I work for digium. fwiw - I use a digium phone and I'm a huge fan. I use the contacts app daily. Love being able to see if someone's already on the phone before I give them a call. |
| 16:00.12 | roe | bchia, to be honest, the only reason I am looking at them is because we're looking at deploying a large asteriskNow-based pbx and we're interested in the phone provisioning feature |
| 16:00.16 | bchia | (I'm also curious if there are any non-digiumites in the channels using digium phones and what they think.) |
| 16:01.23 | bchia | yeah - the asteriskNow provisioning is crazy simple |
| 16:01.37 | roe | and I've heard mixed success with polycom phones |
| 16:03.08 | roe | bchia, do you guys provide samples for testing purposes? |
| 16:05.35 | Qwell | roe: we don't directly, but some reseller/integrator might |
| 16:05.49 | Qwell | I kinda doubt it though. That doesn't sound very common. |
| 16:06.13 | *** join/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 16:06.20 | bchia | Like sample phones? I don't think so, but they d40's a super cheap to pick up 1 or 2 to test with. |
| 16:09.09 | *** join/#asterisk coppice (~chatzilla@123203240103.ctinets.com) |
| 16:13.24 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:13.39 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-193.mobile.uci.edu) |
| 16:14.23 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-193.mobile.uci.edu) |
| 16:14.45 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 16:15.09 | *** join/#asterisk jsjc (~Adium@206.Red-2-136-102.dynamicIP.rima-tde.net) |
| 16:15.56 | *** join/#asterisk cusco_ (~tralala@a79-168-182-209.cpe.netcabo.pt) |
| 16:16.20 | cusco_ | hi |
| 16:17.05 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:17.40 | cusco_ | Im trying to set up cdr via freetds |
| 16:18.42 | Qwell | cusco_: I'm sorry to hear that. |
| 16:18.55 | cusco_ | so am I, believe me |
| 16:19.38 | cusco_ | for ages I been trying to make them believe that mysql is the only choice (actually using) or others suchs as pgsql or sqlite |
| 16:19.56 | cusco_ | but time as come as they need some stuff on mssql... |
| 16:19.58 | cusco_ | :( |
| 16:20.19 | cusco_ | anyways |
| 16:20.31 | cusco_ | I think I have cdr_tds.conf set up |
| 16:20.40 | cusco_ | but might have a problem with freetds.conf |
| 16:20.45 | *** join/#asterisk oej_ (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 16:24.27 | stegbth | WIMPy: thank you very much |
| 16:24.32 | stegbth | it worked |
| 16:25.50 | *** join/#asterisk irroot (~gregory@2002:2935:7f5::1) |
| 16:26.25 | *** part/#asterisk irroot (~gregory@2002:2935:7f5::1) |
| 16:27.20 | Sophira | looks at backscroll. |
| 16:27.25 | Sophira | 15:58 < WIMPy> SIP wasn't made for telephony and that's still pretty obvious. |
| 16:27.31 | Sophira | Interesting, I thought it was. |
| 16:30.01 | *** part/#asterisk stegbth (~stegbth@2001:6f8:9fa:0:221:70ff:fea9:b5e2) |
| 16:30.26 | coppice | SIP was made for telephony, but by people who didn't understand it |
| 16:31.26 | WIMPy | I'd rather say: Added to SIP. |
| 16:31.45 | coppice | what did SIP do before telephony? |
| 16:31.45 | WIMPy | And I'd prefer to talk about VOIP, not telephony in that context. |
| 16:32.16 | WIMPy | General P2P. |
| 16:33.12 | cusco_ | signaling |
| 16:35.19 | coppice | the original SIP spec was an "oh. H.323 is sooo complex. we can make phone calls simpler than that" spec, except it ended up a lot more complex |
| 16:35.55 | Qwell | cusco_: Why not use ODBC? |
| 16:36.26 | WIMPy | Are you sure VOIP was already part of the intention for SIP? I thought it was planned as a general purpose thing, like e.g. online games. |
| 16:39.01 | *** join/#asterisk wonderworld (~ww@dsdf-4db53358.pool.mediaWays.net) |
| 16:39.39 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:40.40 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 16:41.06 | coppice | it was never intended for games. trying looking at the oldest SIP spec |
| 16:41.06 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:41.50 | WIMPy | Just an example for everything. |
| 16:41.51 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 16:42.28 | WIMPy | Or a replacement for teamviewer or whatever. |
| 16:43.09 | jpsharp | I used to think H323 was made by telephony guys with no idea of networking and SIP was made by networking guys with no concept of telephony. Now I think SIP was just made by someone with their head up their ass. |
| 16:43.14 | WIMPy | The locatipn part that is, obviousely. |
| 16:44.03 | coppice | WIMPy: how would SIP be a replacement for something which came later? |
| 16:44.11 | WIMPy | jpsharp: Yes, kind of. But I don't see where H323 is less internet friendly than SIP. They both use RTP. |
| 16:44.47 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 16:44.57 | WIMPy | coppice: Just examples for "general purpose". |
| 16:45.19 | coppice | H.323 was build on decades of experience with telephony. SIP was build by a couple of half wits in a university |
| 16:45.50 | *** join/#asterisk navaismo (~navaismo@189.144.212.238) |
| 16:45.52 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 16:46.02 | jpsharp | WIMPy: I'm not saying H323 is internet friendly at all. |
| 16:47.09 | jpsharp | If only Mark had gotten IAX out before SIP. |
| 16:47.25 | WIMPy | jpsharp: Yes, buy why is it less internet friendly than SIP? |
| 16:47.35 | coppice | its a pity SCTP wasn't available earlier. the development of these signaling protocols could have been a lot different |
| 16:47.50 | coppice | jpsharp: iax is a dead end |
| 16:47.59 | WIMPy | If only IAX would receive a littel more attention these days. It has an extremely better concept. |
| 16:48.07 | *** join/#asterisk camerin (hoax@newelite.bshellz.net) |
| 16:48.23 | coppice | IAX is a bad concept. it merges signaling and media |
| 16:48.49 | WIMPy | Why is that bad? |
| 16:49.21 | coppice | because the media and signaling do not generally want to take the same path |
| 16:49.23 | WIMPy | And I think IAXs signalling is a good compromise between H323 and SIP. |
| 16:49.37 | WIMPy | But they don't have to. |
| 16:50.10 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-kqtgnnoxxslixhln) |
| 16:50.29 | jpsharp | SIP would have been better if the endpoint IPs weren't carried in the message and just let the endpoints send data to the IP source address. |
| 16:56.00 | *** join/#asterisk Galen (~Galen@rrcs-24-43-20-118.west.biz.rr.com) |
| 16:58.05 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 17:02.40 | autofsckk | hello, i want to work with the /etc directory from one server, and im mounting it via sshfs but with a normal user, do i have to be root so i can modify files on the remote server? or is there a specific way to let normal user modify files with root needs? |
| 17:06.24 | *** join/#asterisk gvm (~chatzilla@cpc1-cmbg3-0-0-cust871.5-4.cable.virginmedia.com) |
| 17:07.32 | wonderworld | without NAT there would be worldwide freedom of voice communication. maybe ipv6 will improve things... as long as proxies are needed, it's not decentralized enough. |
| 17:08.38 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 17:11.06 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 17:11.44 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 17:15.24 | coppice | most of the world's blocks on voice communication are not due to NAT |
| 17:16.00 | acidfoo | coppice, there are due to what |
| 17:16.12 | raden | how can i set my callerid to random DID's |
| 17:16.36 | wonderworld | true, but a different culture of contactng people would arise if every peer could reach every other peer encrypted, instantly and without needing 3rd parties |
| 17:17.14 | coppice | acidfoo: companies that don't want to allow direct communications from their employee's machines and countries trying to restrict things for commercial reasons |
| 17:17.51 | WIMPy | I think the political reasons have become even stronger than the commercial ones. |
| 17:20.21 | wonderworld | i am really wondering how they will control encryprion in the future. they can't make it unlawful, because it's clearly needed for many reasons, but they don't want the masses to have it as well. especially not with ipv6. |
| 17:20.26 | coppice | what many perceive as political are actually commercial, like a government collected huge amounts to provide a franchise, and if that's undermined the companies will demand their money back |
| 17:21.37 | WIMPy | if some govenments pay ITSPs all expenses so that they don't set up direct media any more that's surely not for commercial reasons. |
| 17:22.15 | acidfoo | wonderworld, what they might do is to throttle all the encrypted data to a rate that it is almost useless... to push you to not encrypt your data |
| 17:22.28 | wonderworld | acidfoo: yes, might be |
| 17:22.42 | acidfoo | I see no other way, i'd be curious what else they could do |
| 17:22.53 | wonderworld | maybe increase the cost for encrypted traffic |
| 17:22.55 | WIMPy | Or you have to pay extra for encrypted traffic. |
| 17:23.02 | acidfoo | yeah |
| 17:23.16 | acidfoo | if they detect some encrypted traffic they charge you :P |
| 17:23.32 | zombu2 | if |
| 17:23.41 | acidfoo | this is where some stenography breakthrough will emerge ;) |
| 17:23.59 | wonderworld | yes, send sip/rdp thru porn movies |
| 17:24.10 | acidfoo | ;) |
| 17:24.18 | acidfoo | there we are |
| 17:24.42 | WIMPy | So traffic will increase immensely when we transfer lots of junkt to hide the messages. |
| 17:24.49 | *** join/#asterisk cusco (~tralala@a79-168-182-209.cpe.netcabo.pt) |
| 17:24.50 | zombu2 | why encrypt the traffic if you can scramble the voice in the phone then send it plain |
| 17:24.52 | cusco | hmm |
| 17:25.08 | cusco | can I in cdr_tds.conf set aliases? |
| 17:25.26 | acidfoo | zombu2, because they will just unscramble it ? |
| 17:25.32 | cusco | alias UID => uid |
| 17:26.12 | zombu2 | hmm |
| 17:26.29 | wonderworld | but if they would decrease the quality or increase the cost of encrypted traffic, what do they do with https? |
| 17:26.51 | acidfoo | same thing |
| 17:26.59 | acidfoo | and ask google to pay ;) |
| 17:27.02 | WIMPy | Reverse charging for online shops? |
| 17:27.16 | zombu2 | we could make our own internet with blackjack and hookers |
| 17:27.22 | acidfoo | they'll ask the "service" to pay because we access the service in an encrypted way |
| 17:27.23 | wonderworld | maybe whitelists..... |
| 17:27.33 | wonderworld | "lawful" encryption for registered domains |
| 17:27.43 | acidfoo | (this is just a guess hehe) |
| 17:28.39 | WIMPy | We don;t need encryption any more then. If the whole internet is controlled byt the government, it will be safe. |
| 17:28.51 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-vcggonibicyjtgcw) |
| 17:28.56 | wonderworld | ha! you are right. how could i miss that point :) |
| 17:28.58 | acidfoo | WIMPy, AH! |
| 17:29.14 | acidfoo | of course, they are ... what's the word... security expert ! |
| 17:29.39 | lorsungcu | yeah i guess you could argue that the NSA is somewhat of a security expert. |
| 17:30.07 | wonderworld | yes, NSA should run all things |
| 17:30.10 | wonderworld | would be best |
| 17:30.34 | zombu2 | i should run stuff and i garantee that there will be no garantees |
| 17:39.19 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:40.46 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:49.57 | WIMPy | Why does the Astribank 4 PRI have 8 connectors? |
| 17:50.56 | *** join/#asterisk jsjc (~Adium@206.Red-2-136-102.dynamicIP.rima-tde.net) |
| 17:52.04 | tzafrir_laptop | WIMPy, for each port you have two connectors |
| 17:52.14 | Qwell | tzafrir_laptop: Why is that? |
| 17:52.18 | tzafrir_laptop | The top one is crossed |
| 17:52.32 | Qwell | so 1 is straight-through, the other is cross-over? |
| 17:52.38 | tzafrir_laptop | yes |
| 17:52.45 | Qwell | hmm, okay |
| 17:52.48 | WIMPy | That makes sense. But the labeling does not. |
| 17:53.12 | WIMPy | I see both NT and TE both on top and at the bottom. |
| 17:53.36 | WIMPy | And why are there no downloadable manuals? |
| 17:55.35 | WIMPy | tzafrir_laptop: Do you know if it's possible to put the 4 PRI thing in to the mini case the 16 FXO is sold in? |
| 17:55.44 | n3hxs | NT is crossover |
| 17:55.51 | n3hxs | TE is straight |
| 17:56.11 | n3hxs | So when using my test set, I use TE to connect to the NIU. |
| 17:56.47 | tzafrir_laptop | WIMPy, no. Only a 19" is usd (also for FXOs and such) |
| 17:57.11 | tzafrir_laptop | yeah. The labeling of NT and TE is slightly misleading |
| 17:57.40 | WIMPy | The labeling looks like the LEDs would indicate NT or TE mode. |
| 18:03.56 | talntidd | anyone here want a sangoma a101? $250! :) |
| 18:05.17 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 18:05.40 | WIMPy | Maybe you should check ebay for prices of used pri cards. |
| 18:05.49 | talntidd | do'h |
| 18:06.08 | WIMPy | Ah, you just did? ;-) |
| 18:06.36 | talntidd | http://www.ebay.com/sch/i.html?_nkw=sangoma+a101d&_sacat=0&_odkw=sangoma+a101&_osacat=0 |
| 18:07.22 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 18:07.37 | talntidd | those are showing pretty expensive |
| 18:07.49 | talntidd | I don't see any used ones that match |
| 18:08.14 | *** join/#asterisk indika1978 (~chatzilla@61.245.163.62) |
| 18:08.42 | WIMPy | Yes, used Sangoma is expensive, but I'm not sure there's a reason to pay more than for a Digium one. |
| 18:09.55 | wonderworld | how much is a T1 patton? |
| 18:10.19 | [TK]D-Fender | That is not disproportionately expensiv |
| 18:11.00 | wonderworld | http://www.ebay.de/itm/PATTON-SmartNode-4960-Multi-Port-T1-E1-PRI-VoIP-Router-/150884043151?pt=DE_Computer_VoIP_Telefone&hash=item232163c98f |
| 18:11.11 | wonderworld | used EUR 159....without all the pci-hassle |
| 18:12.37 | *** join/#asterisk Tim_Toady (~fuzzy@188.4.77.140.dsl.dyn.forthnet.gr) |
| 18:13.45 | talntidd | so $250 is a bad price, ya think, WIMPy? |
| 18:14.03 | [TK]D-Fender | Sangoma's warranty was a lot longer, and were more reusable as they were 3.3/5v compatible unlike their Digium counterparts |
| 18:14.14 | [TK]D-Fender | $250 isn't bad |
| 18:14.23 | WIMPy | That's a lot more than I have given for 4 port Digium cards. |
| 18:14.50 | talntidd | WIMPy, because a saturn gets from A-B, doesn't mean a mercedes should be priced at saturn levels :) |
| 18:14.50 | [TK]D-Fender | Maybe that tells you something by itself |
| 18:15.03 | [TK]D-Fender | Carzy sellers, or the cards have issues, etc... |
| 18:15.04 | WIMPy | Yes, that either 5V or 3.3V thing is a litte bad. |
| 18:15.56 | *** join/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net) |
| 18:16.19 | [TK]D-Fender | Sangomas are also much better on IRQ resources, support more OSs, better data modes, etc. The scope covers more. |
| 18:16.22 | indika1978 | my dial plan is call screening one. If person x originate a call to pbx and it receives, specified ring group(extensin ,A,B,c)of the pbx is set to ringing. Any one of the group lift the handset, let say A, it gets 2 option, 1.to bridge call X and A, 2. get the x's cid annoucnement.I did this and requirement change, once call came from x to pbx, ring group keep riginging, and if A and B lift... |
| 18:16.24 | indika1978 | ...the handset, both need to have option 1.bridge call to X 2.listen CID of x. Second dial plan is it is really difficult me to start. How can i write 2 nd one? |
| 18:17.09 | WIMPy | Well, Sangoma ends up using DAHDI as well, so I would expect it to have at least the same issues. And As far as I know they have the same issue with only one timing source. |
| 18:17.14 | rbowles | hello, new to this IRC be gentle, I'm testing Asterisk 11 and having all my SIP registers 'disappear' over time ... where should I look for existing/known bug and/or report it? |
| 18:17.57 | WIMPy | rbowles: What does "disappear" mean? |
| 18:18.11 | rbowles | sip show registry is empty |
| 18:18.19 | rbowles | restart fixes it for a few days |
| 18:18.56 | rbowles | i'm registering to: a local SER, a local instance of Yate, remote SERs and some SMB SIP providers |
| 18:19.00 | kaldemar | rbowles: the beta is not stable, it has issues with registers. |
| 18:19.07 | WIMPy | rbowles: Yes, I know that one. I'm pretty sure it's caused by some old, no longer supported configuration keyword somehwere in sip.conf. |
| 18:19.30 | rbowles | ah, i did some cleansing of my sip.conf but perhaps not enough :-) |
| 18:19.34 | WIMPy | I've had that issue *long* before 11. |
| 18:19.40 | kaldemar | rbowles: use an older version until those get sorted out. |
| 18:20.01 | rbowles | ack, if they are known, I'll chill at 10.last until beta2 of 11 :-) |
| 18:20.04 | WIMPy | I fixed it by using callbackexten instead of register. |
| 18:20.38 | WIMPy | I'm pretty sure I've seen that issue way back in 1.8. |
| 18:20.54 | [TK]D-Fender | WIMPy, DAHDI is a raw interface level, it isn't the core card drive however. Wanpipe "deposits" packets into it. |
| 18:21.02 | WIMPy | The registrations disappear when doing 'sip reload'. |
| 18:21.19 | rbowles | this config was rock solid with 1.4/1.6/1.8/10 ... sip reload has no effect, restart fixes |
| 18:21.44 | rbowles | this isn't production config, just my personal box that is usually running beta or RC |
| 18:21.47 | WIMPy | For me 'sip reload' breaks it. Restart fixes it. |
| 18:22.20 | indika1978 | Hi all, How do we take multiple calls in same dialplan / |
| 18:22.52 | WIMPy | indika1978: Your dialplan doesn't care how many calls you get. |
| 18:24.03 | indika1978 | WIMPy pls read by original request which post previous to this. |
| 18:24.35 | WIMPy | indika1978: I did some days ago. Did you read the book, yet? |
| 18:25.09 | *** join/#asterisk d0pe_ (~d0pe_@31-16-49-111-dynip.superkabel.de) |
| 18:25.35 | WIMPy | indika1978: But to give you a pointer: Use local channels. |
| 18:26.15 | d0pe_ | ATA - GrandStream HT503 - self signed certificate (from contrib script) not working, they work perfectly using Blink and ATA works perfectly without TLS |
| 18:26.44 | d0pe_ | btw , hi :) , I love Asterisk |
| 18:27.21 | indika1978 | Yes thaks for that, but local channles means ... ok How can I read the specific parts.. pls give me little bit more on this to go through it. |
| 18:27.23 | talntidd | have you bought the asterisk book? |
| 18:27.27 | talntidd | d0pe_ ^ |
| 18:28.04 | WIMPy | indika1978: Make the "group" of local channels that each call one phone. |
| 18:29.10 | indika1978 | ok thats good thanks .. now I can think .. thanks for 1000 times |
| 18:31.21 | d0pe_ | nope, got quite well trough for now with howtos on the net and my friends / colleges advices for now, it the any specific book that would help me get trough this problem ? |
| 18:32.12 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 18:32.51 | *** join/#asterisk n0tk (~n0tk@gateway/tor-sasl/n0tk) |
| 18:49.57 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 18:52.09 | *** join/#asterisk bipolar (~bipolar@204.186.46.94) |
| 18:54.48 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-aibphvmzjdpklxdq) |
| 18:54.49 | ectospasm | People still use Trillian? Wow |
| 18:55.15 | fenrus | trillian. lol. |
| 18:57.08 | rbowles | pidgin all the way :-) |
| 18:58.33 | *** join/#asterisk gboudreau (~gboudreau@mail2.danslereseau.com) |
| 18:59.37 | gboudreau | in a extension.conf context, if I have extension 2 defined as the Directory, and extension _1XX defined as Dial ${EXT}, and I try to dial 120, I get the Directory. |
| 19:00.15 | gboudreau | how can I ask Asterisk to wait the digit timeout # of seconds, before going to the Directory (i.e. prioritize my _1XX extension over the 2 extension) |
| 19:03.35 | Qwell | gboudreau: those would never match the same thing. |
| 19:03.44 | Qwell | If it's doing that, you've done something wrong. |
| 19:03.54 | *** part/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net) |
| 19:03.56 | gboudreau | as soon as I hit 2, of the 1 2 0, I get the Directory. |
| 19:04.03 | [TK]D-Fender | gboudreau, PASTEBIN the whole thing |
| 19:04.05 | [TK]D-Fender | ~pb |
| 19:04.05 | 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. |
| 19:04.05 | gboudreau | ok, let me paste my dialplan show. |
| 19:04.06 | [TK]D-Fender | ^^^ |
| 19:04.19 | Qwell | Then your 1 is being missed or eaten somehow. |
| 19:04.41 | gboudreau | http://pastebin.com/raw.php?i=vpSmFZ4S |
| 19:04.51 | gboudreau | my _1XX appears at the end. not sure this is relevant. |
| 19:05.01 | gboudreau | It's not at the end of my conf file, but I don't think that matters. |
| 19:05.15 | [TK]D-Fender | it isn't. If that is being used as it appears then the 1 is getting lost entirely somehow |
| 19:05.33 | *** join/#asterisk Simnol (~Adam@cpc2-stre5-2-0-cust104.1-1.cable.virginmedia.com) |
| 19:05.47 | Qwell | Show the dialplan execution from the call. |
| 19:05.57 | Qwell | I'm sure the i is getting hit. |
| 19:06.56 | gboudreau | http://pastebin.com/raw.php?i=50PGCcye |
| 19:07.11 | gboudreau | how can I see the # that asterisk receives ? |
| 19:07.17 | Qwell | What's before that? |
| 19:07.38 | gboudreau | coming from another context |
| 19:07.43 | Qwell | show the whole call |
| 19:08.07 | gboudreau | http://pastebin.com/raw.php?i=6JNhjEgZ |
| 19:08.56 | [TK]D-Fender | gboudreau, [ Context 'main-menu' created by 'pbx_config' ] |
| 19:09.04 | [TK]D-Fender | gboudreau, Executing [s@dial-in-menu:1] Answer("SIP/120-00000003", "") in new stack |
| 19:09.12 | Qwell | [TK]D-Fender: There's a goto |
| 19:09.48 | [TK]D-Fender | just caught that... |
| 19:09.57 | [TK]D-Fender | maybe we didn't need teh ENTIRE call :) |
| 19:10.05 | Qwell | I don't see how you're getting from s@dial-in-menu:6 to 2@dial-in-menu:1 though. |
| 19:10.41 | Qwell | wait, of course I do. |
| 19:10.45 | Qwell | You're using Background(,m) |
| 19:10.47 | Qwell | Don't do that. |
| 19:11.06 | Qwell | There is no extension 1, so it ignores it. You haven't even gotten to the WaitExten() |
| 19:11.31 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 19:11.47 | Qwell | <PROTECTED> |
| 19:11.57 | Qwell | Why Background actually has that option is beyond me though. |
| 19:11.57 | gboudreau | good catch :) fixed indeed. |
| 19:12.07 | gboudreau | I copy-pasted that line from somewhere else. meh @ me. |
| 19:12.36 | [TK]D-Fender | gboudreau, congratulations ... that's the first time I've every seen that parameter used. In over 7 years. |
| 19:12.44 | gboudreau | eheh |
| 19:12.51 | [TK]D-Fender | gboudreau, Never ever do that again :p |
| 19:13.08 | [TK]D-Fender | passed my radar, that's for sure |
| 19:13.20 | Qwell | [TK]D-Fender: Admittedly, that option makes no damn sense. |
| 19:13.29 | gboudreau | it makes the menu more responsive. as soon as the user hits 1 or 2, it goes to the next context, instead of having to wait whatever digit timeout I might have defined. |
| 19:13.35 | Qwell | it's like...don't actually Background. What, no, screw you app_background. |
| 19:14.14 | [TK]D-Fender | gboudreau, that is not a healthy command |
| 19:14.22 | Qwell | gboudreau: You would use Read() for that instead. |
| 19:14.32 | gboudreau | ah, ok. will do. |
| 19:14.47 | Qwell | that or don't put patterns in that context |
| 19:17.19 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 19:17.33 | gboudreau | well, you guys are useful :) I might be back later on. I'm almost done with my setup. just need to configure follow-me with start and end times during weekdays... |
| 19:29.00 | Katty | sighs |
| 19:29.40 | Qwell | no sighing allowed. :( |
| 19:33.54 | ectospasm | was that a sigh or a swoon? |
| 19:34.21 | talntidd | textually, it was a sigh. |
| 19:34.36 | talntidd | very difficult to typo "sigh" while attempting to type "swoon" |
| 19:35.11 | ectospasm | talntidd: but you *can* sigh as you swoon, no? |
| 19:35.19 | Katty | no, no it's not a swoon |
| 19:35.24 | Katty | it's an irritated sigh. |
| 19:35.33 | ectospasm | and the only authoritative voice here speaks again! |
| 19:36.11 | Katty | we just put in a new appliance that tracks packets going on the wan port. |
| 19:36.20 | Katty | s/on/through/ |
| 19:36.31 | Katty | and it seems that someone is quite regularly downloading things from Steam |
| 19:36.38 | talntidd | sorry 'bout that. |
| 19:36.43 | talntidd | ill stop. |
| 19:37.27 | Katty | unfortunately i don't get to play network gestapo |
| 19:37.30 | Katty | so there's nothing i can do about it |
| 19:37.47 | talntidd | ipchains -I DROP |
| 19:37.48 | talntidd | ... |
| 19:38.06 | ectospasm | Katty: sounds like your organization lacks a strong AUP |
| 19:38.15 | Katty | i don't even know what aup stands for. |
| 19:38.26 | Katty | so that's probably a fair statement |
| 19:38.32 | zombu2 | or someone that gets a kick out of dropping packets |
| 19:39.03 | Katty | people already dislike me. people think IT folk are lazy |
| 19:39.21 | Katty | i'm not about to start playing Network Police |
| 19:39.36 | Katty | so i keep notifying the owner |
| 19:39.42 | Katty | of course he does nothing. |
| 19:39.54 | zombu2 | hmm would you get up and push a button if you can write a quick prog that does it for you |
| 19:39.57 | Katty | i guess maybe he just doesn't care. |
| 19:43.08 | talntidd | or you are not explaining why it is bad very well |
| 19:43.26 | talntidd | owners have lots of shit to do every day.. they need bottom line of why/what is bad |
| 19:43.34 | talntidd | not "X is doing Y" |
| 19:44.32 | zombu2 | or maybe you shouldn t even bother with it since it is really not your problem |
| 19:44.53 | zombu2 | if it becomes a prob for the owner he will act on it s own |
| 19:52.23 | Katty | in case you weren't aware. i was a venting |
| 19:52.27 | Katty | as females do |
| 19:52.30 | Katty | i do not need to be fixed. |
| 19:52.41 | Katty | but thank you for trying, regardless (= |
| 19:52.59 | *** part/#asterisk gboudreau (~gboudreau@mail2.danslereseau.com) |
| 19:54.32 | ectospasm | Katty: AUP - Acceptable Use Policy |
| 19:54.40 | ectospasm | ...what IT should be enforcing |
| 19:55.03 | talntidd | fuck that |
| 19:55.09 | talntidd | IT shouldn't enforce shit |
| 19:55.14 | talntidd | the management should enforce it |
| 19:55.21 | talntidd | IT should just give them the tools to manage it |
| 19:55.37 | talntidd | and the ability to see who is breaking it |
| 19:55.53 | talntidd | at my call center, I have to fire people when they break our AUP |
| 19:56.02 | talntidd | that wasn't in my job description. lol |
| 19:56.48 | talntidd | I don't wanna hear peoples "I have 2 kids at home and am a single mother! I didn't know that proxying around your blocks and playing facebook games all day was against policies!" |
| 19:56.50 | Katty | i'd would just flat out refuse to do that. |
| 19:57.15 | talntidd | I came up with a creative way to let the problem work itself out |
| 19:57.34 | talntidd | deluser <username>; ssh root@thinclientserver; slay <username> |
| 19:57.52 | talntidd | then when they don't set any appointments, it becomes a management issue! |
| 19:59.48 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 20:04.44 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 20:05.28 | ectospasm | talntidd: well, it is management that approves the AUP |
| 20:05.35 | ectospasm | it's up to them to decide who manages it |
| 20:05.56 | ectospasm | or who enforces it |
| 20:06.29 | ectospasm | depends on what IT defines its role as |
| 20:07.35 | talntidd | ectospasm, generally IT's "role" is modified by whoever gets a stick up their ass. |
| 20:09.00 | talntidd | at one company i worked for, it was the IT dept's "job" to ensure the IT room was always organized and tidy |
| 20:09.01 | *** join/#asterisk d0pe_ (~d0pe_@31-16-49-111-dynip.superkabel.de) |
| 20:09.19 | talntidd | but other staff was instructed to put boxes and such in the IT room for disposal. lol |
| 20:10.11 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 20:11.13 | ectospasm | talntidd: yeah, but IT executives are supposed to define the roles of IT |
| 20:11.38 | talntidd | not in smaller companies. lol |
| 20:11.45 | ectospasm | does IT transform the business? Or do we just support ongoing operations: marketing, production, payroll, finance, etc. |
| 20:12.17 | ectospasm | talntidd: in smaller companies it can be easier, but can can also be more difficult |
| 20:12.23 | ectospasm | depending on the culture |
| 20:14.05 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 20:16.27 | ectospasm | in smaller companies the CIO and CEO can be the same person |
| 20:16.36 | ectospasm | ...therefore making it easier |
| 20:17.52 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 20:18.11 | ectospasm | ...and the role of the CIO is nebulous, too. If they report directly to the CIO, they can effect change possibly more easily, than if they report to the CFO |
| 20:23.48 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:26.29 | [TK]D-Fender | checkout time, later all |
| 20:27.12 | *** join/#asterisk specing_ (~specing@unaffiliated/specing) |
| 20:27.14 | specing_ | Hi |
| 20:27.33 | specing_ | How do you people configure asterisk? |
| 20:27.55 | specing_ | I find plunging through the 12000 lines of config files... time consuming. |
| 20:28.17 | specing_ | using asterisk 1.6.x on debian squeeze |
| 20:28.46 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-ztzqvgeurwecnwds) |
| 20:29.32 | *** join/#asterisk n0tk (~n0tk@gateway/tor-sasl/n0tk) |
| 20:29.46 | specing_ | Is there a GUI available for it out-of-the-box? |
| 20:30.03 | specing_ | I noticed there is some embedded web server |
| 20:31.35 | drmessano | ~book |
| 20:31.35 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:32.15 | specing_ | Oh my, more reading. |
| 20:33.11 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-vaooovcvvojohgsr) |
| 20:33.41 | specing_ | headdesks |
| 20:35.58 | *** join/#asterisk kessius (bd043dde@gateway/web/freenode/ip.189.4.61.222) |
| 20:39.48 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 20:41.44 | *** join/#asterisk jsjc (~Adium@206.Red-2-136-102.dynamicIP.rima-tde.net) |
| 20:45.02 | d0pe_ | anyone had to deal with ATA - Grandstream HT5xx / SIP over TLS ? |
| 20:47.04 | kessius | hi people , I'm newbie to asterisk , how to asterisk recognition voice |
| 20:47.17 | wonderworld | specing_: http://www.voip-info.org/wiki/view/Asterisk+GUI . but doing some basic configuration isn't that hard. depending on your needs, you should be able to have a working installation running in a few hours. |
| 20:54.36 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:54.47 | kessius | hi, how to implement voice recognition, in asterisk |
| 20:55.13 | [TK]D-Fender | kessius: with some 3rd party app like Sphinx or Lumenvox. |
| 20:55.41 | specing_ | Ok, now Im even more confused |
| 20:55.52 | specing_ | I'll plunge into those 12k lines I guess |
| 20:55.53 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 20:56.25 | wonderworld | specing_: what do you want to do? |
| 20:57.12 | specing_ | call myself when Im out of reach. |
| 20:57.51 | kessius | [TK]D-Fender thanks |
| 20:58.09 | *** join/#asterisk italorossi (~Adium@189.124.200.92) |
| 21:02.22 | specing_ | Is there a recommended softphone I can use to test my misconfiguration? |
| 21:02.57 | [TK]D-Fender | specing_: Any should do |
| 21:03.03 | [TK]D-Fender | ~softphone |
| 21:03.03 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 21:03.05 | Chainsaw | specing_: Main thing to keep in mind is that Asterisk isn't a phone system. It's a toolkit to construct one. |
| 21:03.51 | Chainsaw | specing_: The book is good, if only to teach you the terminology that allows you to ask the right questions. |
| 21:09.59 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-fyrtuuajudtyajlf) |
| 21:15.28 | cusco | ~bria |
| 21:15.28 | infobot | [~bria] Bria is a NON-free SIP soft-phone for Windows, Mac OS X, and Linux that can be downloaded from http://www.counterpath.com . This includes licensed audio & video codecs and is full-featured. |
| 21:15.43 | cusco | <PROTECTED> |
| 21:15.43 | infobot | [~eyebeam] eyeBeam is a NON-free SIP soft-phone for Windows, Mac OS X, and Linux that can be downloaded from http://www.counterpath.com . This includes licensed audio & video codecs and is full-featured. |
| 21:15.52 | cusco | o.O |
| 21:20.11 | [TK]D-Fender | What is matter? |
| 21:20.42 | cusco | what is the difference? |
| 21:23.09 | [TK]D-Fender | interface and functin. |
| 21:23.16 | [TK]D-Fender | integration and so forth |
| 21:25.52 | cusco | ok.. |
| 21:26.10 | cusco | so.. ws support in asterisk sounds promissing |
| 21:29.57 | drmessano | ws? |
| 21:30.23 | drmessano | Oh NM |
| 21:33.37 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 21:40.13 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 21:43.53 | specing_ | <3 lua |
| 22:00.40 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 22:05.16 | *** join/#asterisk Tarso (~Cookie_Mo@189.61.47.149) |
| 22:06.11 | *** join/#asterisk darkdrgn2k (~darkdrgn2@69-165-131-20.dsl.teksavvy.com) |
| 22:06.12 | darkdrgn2k | hey all |
| 22:06.31 | darkdrgn2k | any one know how to specify the provisiojng server on a 1120e avaya/nortel sip phone? |
| 22:07.19 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 22:07.32 | talntidd | https://www.google.com/search?q=provisioning+server+1120e&sugexp=chrome,mod=7&sourceid=chrome&ie=UTF-8 |
| 22:07.38 | talntidd | http://track.sipfoundry.org/secure/attachment/26591/NN43112-300_04.02_SIP2.2_1120E_Admin.pdf |
| 22:07.53 | darkdrgn2k | lol actualy its the same doc im looking at |
| 22:08.12 | darkdrgn2k | ... seems you nee dto put the phone into FULL DHCP mode.. |
| 22:08.17 | darkdrgn2k | hmm wonder what that means |
| 22:09.23 | talntidd | that doc makes it seem pretty straightforward |
| 22:09.41 | talntidd | starting on page 25 |
| 22:09.43 | darkdrgn2k | yeh im reading im reading... |
| 22:10.23 | darkdrgn2k | "The DHCP server recognizes this |
| 22:10.23 | darkdrgn2k | special Vendor Class Identifier (Option 60) and sends back OFFER, which |
| 22:10.23 | darkdrgn2k | also includes the same Vendor Class Identifier" |
| 22:10.26 | darkdrgn2k | huh |
| 22:15.08 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:31.10 | talntidd | darkdrgn2k it means... |
| 22:31.18 | talntidd | you have to let the phone get its IP from DHCP |
| 22:31.38 | talntidd | then, in your DHCP server, setup option 60, to serve the address of the server that stores the .cfg file for the phones |
| 22:32.14 | talntidd | so when the phone boots, and gets its IP, the DHCP server also serves the address for Option 60 |
| 22:32.37 | talntidd | what kind of dhcp server you running? |
| 22:41.26 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-xxdywjxifkqcpciy) |
| 22:41.35 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-dxzhmcgxtvocmhyj) |
| 22:44.38 | darkdrgn2k | Right now im using DNSMASQ but in produiction it will be cisco |
| 22:44.58 | darkdrgn2k | i alreayd have 66 and 150 defined as the provisiongin server but that doesnt work |
| 22:45.22 | darkdrgn2k | if i set the provisioning server manually it works find |
| 22:45.28 | darkdrgn2k | bu ti need to find a way to do it via dhcp |
| 22:48.56 | jpsharp | I'd sniff the network to make sure your dhcp server is sending out the data. |
| 22:49.25 | darkdrgn2k | it is.. its some sutpid Vedor ident string or somethign the phone is expecting |
| 22:49.27 | darkdrgn2k | dump nortel |
| 22:51.30 | talntidd | :) |
| 22:51.32 | jpsharp | Oh, look. There was a problem on the POE switch that these phones were connected to and fried them all. Guess you'll need to buy new phones. |
| 22:53.12 | talntidd | I had that happen with some some cisco switches and 92 phones. |
| 22:53.33 | talntidd | the phones would boot, and function, but it was very staticy.... |
| 22:53.39 | talntidd | took forever to troubleshoot it. |
| 22:54.15 | talntidd | replaced a phone, and it was staticy.. replaced switch.. phones still static... |
| 22:54.25 | talntidd | replaced phone again on new switch, phone worked properly |
| 22:54.34 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 22:54.38 | talntidd | when i replaced the original phone, it instantly got fried from the bad switch. |
| 22:54.43 | talntidd | same way the others were |
| 22:56.44 | jpsharp | Sounds like bad filtering in the POE circuit. |
| 22:57.05 | talntidd | i still have the switch.. |
| 22:57.13 | talntidd | some day i'll dig in and see what happaned |
| 22:57.16 | talntidd | happened |
| 22:57.18 | talntidd | but it sucked. |
| 22:57.25 | talntidd | next day'd 95 phones |
| 22:57.30 | talntidd | and 4 switches |
| 22:57.38 | talntidd | corporate loved that bill. :) |
| 22:57.40 | darkdrgn2k | <talntidd>: wierd i got a bunch of phones that are staticy |
| 22:57.43 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 23:02.30 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 23:33.10 | *** part/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 23:40.35 | phix | talntidd: You should test each port with your tounge, should give you a good indication which one it was :) |
| 23:42.06 | *** part/#asterisk Dar1us (~doconnor@cain.gsoft.com.au) |
| 23:43.21 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-ljkpkftxcdggqmiw) |
| 23:50.38 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 23:50.50 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |