| 00:01.39 | ChannelZ | riwarren: looks like you have authentication issues |
| 00:02.11 | *** join/#asterisk HyperNerdV2 (~HyperNerd@rrcs-24-43-162-198.west.biz.rr.com) |
| 00:12.53 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
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| 00:26.44 | wonderworld | when i call asterisk from any SIP-client the first Playback() in the dialplan stutters very badly. after that everything is fine. any idea on how to fix this? |
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| 01:15.06 | mazpe | is there neat trick of some kind that will allow me to have same extension number on two servers that are link via sip? |
| 01:15.26 | mazpe | ext 105 on server a -- exten 105 on server b |
| 01:15.50 | mazpe | the idea is to dial using a prefix... 1 to dial server a and 2 to dial server b.. |
| 01:17.15 | mazpe | so dialing from 105 on server B from server A would be 2105... then on serverA i will ${EXTEN:1}@serverB |
| 01:17.49 | [TK]D-Fender | Yes.. that is your DIAL command |
| 01:18.09 | mazpe | this works... but not really.. i get "username mismatch, have <105>, digest has <serverA> |
| 01:18.13 | [TK]D-Fender | What you dial from one box too the other is a manipulation of the number you pass in the Dial(). You can send them whatever you want |
| 01:18.25 | [TK]D-Fender | And you don't dial from an "extension". |
| 01:18.36 | [TK]D-Fender | You dial from a device and that is a number in your dialplan. |
| 01:18.42 | mazpe | right.. |
| 01:19.03 | mazpe | and that works... for some reason if i have a device with same name... it seems to trip. |
| 01:19.47 | mazpe | oh wait.. i could [serverA-105] regexten=105 right? |
| 01:19.53 | [TK]D-Fender | No. |
| 01:19.58 | [TK]D-Fender | Forget regexten. |
| 01:19.59 | mazpe | no? |
| 01:20.00 | [TK]D-Fender | Forever |
| 01:20.19 | [TK]D-Fender | Not related and is a virtually worthless thing on almost any system |
| 01:20.52 | [TK]D-Fender | Change your DIAL |
| 01:21.17 | mazpe | change my dial? |
| 01:21.38 | [TK]D-Fender | mazpethis works... but not really.. i get "username mismatch, have <105>, digest has <serverA> <- sorry, missed this clarification |
| 01:21.45 | [TK]D-Fender | type = friend |
| 01:21.47 | [TK]D-Fender | ^ |
| 01:21.57 | [TK]D-Fender | it's authing by IP where it was ID'd as another # |
| 01:22.11 | [TK]D-Fender | so you can't jsut auth by IP (peer) |
| 01:22.24 | mazpe | yeah its type = friend |
| 01:22.58 | mazpe | so do a register instead? |
| 01:23.22 | [TK]D-Fender | Shouldn't need. |
| 01:23.35 | [TK]D-Fender | pastebin everything |
| 01:24.03 | mazpe | ok |
| 01:33.14 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 01:34.56 | mazpe | [TK]D-Fender: almost done.. lol |
| 01:38.51 | mazpe | [TK]D-Fender: http://pastebin.com/96DJuVP9 |
| 01:40.09 | [TK]D-Fender | mazpe: for [EMGA] do "fromuser=EMGA", "sendrpid=yes", "trustrpid=yes". Do the same on the other side |
| 01:41.07 | mazpe | doing |
| 01:42.08 | mazpe | it woooorks! |
| 01:42.35 | mazpe | declares [TK]D-Fender pure genius |
| 01:42.37 | mazpe | :) |
| 01:42.50 | [TK]D-Fender | You're welcome |
| 01:43.13 | [TK]D-Fender | it was taking the calleriID when you dialed out that peer and trying to match the username which happened to match the username on the other system. |
| 01:43.29 | [TK]D-Fender | fromuser+rpid allows you to force the user while maintaining the callerid |
| 01:44.16 | mazpe | makes sense... |
| 01:44.22 | mazpe | sweet. |
| 01:55.12 | *** join/#asterisk TomCat2 (~TomCat@static-72-77-202-50.tampfl.fios.verizon.net) |
| 01:56.52 | TomCat2 | Hi. I'm running 10.6.1 built from sources. I tried to use MeetMe() but Asterisk complained that MeetMe wasn't installed. Is what I'm reading correct in that this feature comes with Dahdi, or am I mistaken? |
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| 01:57.27 | TomCat2 | I just want to create some very basic conference room functionality, so if there's a better route for this, I'm all ears |
| 01:58.31 | [TK]D-Fender | DAHDI is a requirement for MeetMe, yes |
| 01:59.06 | [TK]D-Fender | app_conference does not require it and it's functionality might be close enough for you |
| 01:59.49 | TomCat2 | ok - I'll take a look at that one - having some issues building Dahdi and didn't want to go too far down the road of solving layers of problems where there may be an easier and/or better solution - thanks |
| 02:00.35 | TomCat2 | Application_ConfBridge the one you're referring to? |
| 02:03.09 | TomCat2 | nvm - answered my own question - looks to be what I need - thanks again |
| 02:03.18 | [TK]D-Fender | yes |
| 02:31.19 | *** join/#asterisk MindTheGap (~MindTheGa@177.81.24.119) |
| 02:33.14 | TomCat2 | nice - got confbridge working :D looks nice |
| 02:33.52 | TomCat2 | as always, thanks again [TK]D-Fender |
| 02:35.30 | [TK]D-Fender | You're welcome |
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| 02:38.13 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 06:08.27 | v0lZy | lo |
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| 06:19.32 | schmidts | good morning |
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| 06:36.37 | jpsharp | greetings, program |
| 06:40.46 | schmidts | jpsharp which program? |
| 06:41.34 | jpsharp | cron! |
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| 06:58.23 | v0lZy | hei guys, quick question |
| 06:58.39 | v0lZy | im including some stuff into each phones individual context |
| 06:58.50 | v0lZy | but before inclusion, I want to set some variables |
| 06:59.33 | v0lZy | how should my exten => start in order to proceed with whats included below it? |
| 07:00.23 | v0lZy | i thought about using _X! but then it doesnt continue down the dialplan because the stuff im including starts with a different pattern and priority 1 |
| 07:00.43 | v0lZy | what i need to do, is set something before including something else and then continue down the dialplan |
| 07:02.32 | kaldemar | it does not work that way. |
| 07:02.53 | kaldemar | it the call matches an extension in your contexts, included ones become irrelevant at that point. |
| 07:02.57 | v0lZy | well i know i can add labels and use goto |
| 07:03.10 | v0lZy | ah. |
| 07:03.15 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:03.27 | v0lZy | any way to get what i describe above? |
| 07:03.39 | bulkorok | hi |
| 07:03.44 | kaldemar | either make can overlay context that sets your variables and then does a Goto to a context that has the includes, or use the setvar parameter in sip.conf or global variables. |
| 07:03.44 | v0lZy | basically... set something in context without matching a pattern |
| 07:04.09 | v0lZy | ah, i can setvar in sip.conf |
| 07:04.14 | v0lZy | that would work i think |
| 07:04.28 | kaldemar | you can't do anything in dialplan without a matching extension. period. |
| 07:05.06 | v0lZy | will that be an inherited variable? |
| 07:05.46 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 07:08.09 | kaldemar | if you make it so. |
| 07:08.32 | v0lZy | just prepend the name? |
| 07:08.41 | v0lZy | setvar=__MYVAR=value ? |
| 07:10.47 | kaldemar | yes. |
| 07:10.54 | v0lZy | got it working, thanks |
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| 07:25.44 | *** join/#asterisk jofry (~jofry@port-212-202-54-40.dynamic.qsc.de) |
| 07:25.52 | jofry | hi folks |
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| 07:30.14 | kresp0 | hi jofry |
| 07:35.03 | v0lZy | kaldemar: any other ways to pickup calls apart from pickupmarks? |
| 07:35.32 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 07:39.38 | v0lZy | and another question regarding hints |
| 07:40.08 | v0lZy | if i put it into each individual phones context, nothing will be included ΕΎ |
| 07:40.16 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 07:40.57 | v0lZy | so.. i have to put it towards the end of what im doing, just prior to establishing contact with the application/extension/external number, right? |
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| 07:46.18 | *** join/#asterisk mitt (mitt@173.255.187.227) |
| 07:46.22 | mitt | hi there ... |
| 07:46.30 | mitt | i require some help ... |
| 07:47.09 | v0lZy | shoot |
| 07:47.09 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 07:48.49 | mitt | i am trying to put AMD() on the Incoming SIP Calls |
| 07:49.16 | mitt | but mostly i got the calls which is no Voice or Music on Hold |
| 07:51.59 | mitt | exten => s,1,Answer |
| 07:52.00 | mitt | exten => s,2,AMD() |
| 07:52.00 | mitt | exten => s,n,GoToIf("$[${AMDSTATUS}"="HUMAN"]?humn:mach) |
| 07:52.00 | mitt | exten => s,n(mach),Playback(message-when-machine) |
| 07:52.00 | mitt | exten => s,n,Hangup |
| 07:52.00 | mitt | exten => s,n(humn),Queue(6501,Ttm) |
| 07:52.01 | mitt | exten => s,n,Hangup |
| 07:52.04 | mitt | dialing plan is like this |
| 07:52.28 | mitt | it comes, to server --> connect and tryto check it is human or not , if human want to put it in queue , |
| 07:52.32 | mitt | if not it hangup |
| 07:54.34 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:57.33 | v0lZy | whats aMD? |
| 07:58.14 | mitt | Answering machine detection |
| 07:59.12 | v0lZy | and the above isnt working? |
| 08:03.27 | mitt | yes |
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| 08:12.03 | bulkorok | core show application AMD says: Simply call this application after the call has been answered |
| 08:12.03 | bulkorok | (outbound only, of course). |
| 08:12.30 | bulkorok | you try it with an incomming call!? |
| 08:12.54 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:14.23 | sekil | anyone uses pgsql in dialplan? |
| 08:14.51 | sekil | I see connections on the server not being disconnected...even though disconnect is issued in the dialplan |
| 08:15.44 | bulkorok | did you tried func_odbc yet?! |
| 08:15.52 | sekil | I'm on 1.2 still |
| 08:15.57 | bulkorok | ew |
| 08:15.58 | sekil | is there one on 1.2 |
| 08:16.00 | sekil | ? |
| 08:16.10 | bulkorok | don't think so |
| 08:16.32 | bulkorok | is 1.2 still supported?! |
| 08:16.44 | sekil | I support it on my servers :) |
| 08:16.53 | bulkorok | :) |
| 08:17.20 | bulkorok | how do you connect!? |
| 08:17.25 | bulkorok | to pgsql? |
| 08:17.34 | ChannelZ | His telegraph still works |
| 08:18.18 | sekil | http://www.voip-info.org/wiki/view/Asterisk+PGSQL |
| 08:18.51 | sekil | app_sql_postgres |
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| 08:25.08 | kaldemar | v0lZy: callgroups and pickupgroups. |
| 08:28.00 | v0lZy | hows that different from pickupmarks? |
| 08:28.43 | mitt | yes incoming call, |
| 08:30.06 | kaldemar | v0lZy: in almost every possible way. you just configure callgroups for devices in sip.conf and the ones with the matching pickupgroup can pick up their calls with *8 or something else configured in features.conf. |
| 08:30.50 | v0lZy | that works even without configuring pickupgroups. |
| 08:31.13 | kaldemar | nope. |
| 08:31.47 | mitt | does AMD works on SIP Incoming CALL ? |
| 08:31.59 | v0lZy | kaldemar: it does |
| 08:32.06 | v0lZy | i mean... i set a pickup mark |
| 08:32.13 | v0lZy | and then i can pickup a call with *8 |
| 08:32.19 | v0lZy | i dont have groups configured. |
| 08:34.38 | kaldemar | mitt: is if other that answering machines call you? |
| 08:35.22 | kaldemar | v0lZy: well that's obvious... if you don't have marks nor groups configured, it will not work. |
| 08:36.00 | v0lZy | kaldemar: What im trying to say is, i have only pickupmarks set. I dont have callgroups or pickup groups configured at all and I can still pickup with *8 |
| 08:38.22 | kaldemar | yes, and? |
| 08:38.38 | kaldemar | there is more than one way to do it. |
| 08:40.12 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:40.22 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:44.03 | v0lZy | but... the pickup groups, they dont use the pickupmark mechanism? |
| 08:48.26 | *** join/#asterisk attila_lendvai (~attila_le@unaffiliated/attila-lendvai/x-3126965) |
| 08:48.37 | v0lZy | any special format for playing music on hold? im trying wav but its not working for some reason |
| 08:49.37 | attila_lendvai | hi! any ideas why ast_localtime() returns GMT whatever I do? I pass in NULL explicitly... on the command line 'date' returns the local time properly, but in the C code asterisk seems to insists on GMT whatever I pass in for ast_localtime... this is on an openwrt router if that matters. |
| 08:57.33 | attila_lendvai | google doesn't seem to bring anything helpful... :/ |
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| 09:05.47 | v0lZy | hm |
| 09:05.52 | v0lZy | i cant get custom music on hold working |
| 09:06.00 | v0lZy | i recoded with sox |
| 09:06.06 | v0lZy | but still.. just muted |
| 09:09.08 | v0lZy | ah ok, got it working |
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| 09:38.33 | polomolo747 | hello, i have installed asterisk 10 after i had asterisk 1.8 working(was removed prior to installation), and now i can't execute any sip commands, i didn't delete the original configuration. |
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| 10:02.23 | DanfromUK | Hi. I have to setup BLF on a grandstream gxp2010 phone but not having much luck. Has anyone done this? |
| 10:04.58 | fling | can please someone help me with D-Link DVG-7111S? |
| 10:05.07 | fling | it is not registering with asterisk |
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| 10:07.40 | tzafrir_laptop | polomolo747, what happens when you run (in the asterisk CLI) module load chan_sip.so ? |
| 10:08.56 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
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| 10:45.51 | dym | Is it possible to attach special variables to a callfile to be used in the dialplan later? Say I called Local/$EXTEN@testcontext and ran something there which needed this variable... |
| 10:46.30 | *** join/#asterisk Dennisvj (~dennis@unaffiliated/dennisvj) |
| 10:47.18 | kaldemar | dym: don't know about special but you can use "Setvar: variable=value" to set channel variables. |
| 10:47.27 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files |
| 10:47.50 | dym | sorry kaldemar - i just found that too |
| 10:47.55 | dym | lazy me! |
| 10:49.00 | dym | Can I also use setvar when running an application? |
| 10:49.17 | dym | okay, maybe i should explain from the beginning :D |
| 10:49.32 | *** join/#asterisk darthanubis (~anubis@unaffiliated/darthanubis) |
| 10:52.07 | dym | im trying to interact with an external script. the script writes the callfile to send a fax like this: http://pastebin.com/62xjn5Es - what i need is the external script to get the channelid used to initiate the call, so i thought i let the script insert a uniqueid variable for the call and i return the channelid to match to the script. (this for beeing able to cancel the call) |
| 10:53.02 | dym | But is it still possible to set a variable when connecting to an application? |
| 10:54.01 | wonderworld | when i connect to my asterisk with any SIP-client to an extension with a few Playback()s, the first one always stutters very badly. after that everything is fine. any idea on how to fix that? |
| 10:54.04 | dym | damn that wont work since the variable would only be set if the initiation of the first local call is successful and the fax application is connected |
| 10:55.04 | dym | can i set multiple variables in the callfile? |
| 10:55.47 | wonderworld | dym: yes, you can, done that before, can't remember how |
| 10:58.16 | wonderworld | as a hack you could simply put all your vars into Data like -> Data:/var/spool/asterisk/myfavouritefax.pdf|var2|var3|var4 and split them afterwards in your script |
| 10:58.38 | wonderworld | but i am pretty shure you can pass multiple vars (if i remember correctly) |
| 11:00.04 | dym | wonderworld: ill go for 2 local contexts and use the var definition there |
| 11:00.07 | dym | seems more clean |
| 11:00.10 | dym | but thanks |
| 11:00.11 | dym | (: |
| 11:00.49 | kaldemar | use multiple Setvar lines in the callfile. |
| 11:01.18 | kaldemar | it says so on the wiki page on the line that tells what Setvar does. |
| 11:03.55 | dym | kaldemar, like this: http://pastebin.com/NAHFcqn0 ? |
| 11:04.30 | dym | More than one Setvar: may be specified. |
| 11:04.31 | dym | dammit |
| 11:04.42 | dym | feel free to whack me with random objects |
| 11:05.22 | dym | urgh |
| 11:05.46 | dym | but if i had 2 calls run into that context, that may overwrite the variable too quick before the fax has been processed |
| 11:06.20 | kaldemar | variables are associated with channels, not contexts. |
| 11:06.25 | dym | AH |
| 11:06.43 | dym | a day full of wonders (: |
| 11:13.23 | v0lZy | kaldemar: what does asterisk do if there's a call forwarding loop? |
| 11:13.33 | v0lZy | like A forwards to B and B is forwarding to A |
| 11:14.54 | kaldemar | v0lZy: depends on what A and B are and how the forwarding is done. ask more specific questions please. |
| 11:15.11 | v0lZy | a and b are phones ... internal extensions |
| 11:15.19 | v0lZy | im forwarding with a gotoif check |
| 11:15.37 | v0lZy | i check for a db entry CF/called_extension= |
| 11:15.54 | v0lZy | if its non zero, i forward to whatever the contents are |
| 11:16.16 | v0lZy | just... dial to the value. |
| 11:16.22 | kaldemar | if you do it all in dialplan, then you have a loop that executes applications as fast as your box can. |
| 11:16.39 | v0lZy | how to prevent such a loop? |
| 11:17.01 | v0lZy | i suppose i need to prevent forwarding to forwarded numbers to disable chain forwarding? |
| 11:17.07 | kaldemar | if you already have done it, why are you not testing it for yourself? |
| 11:17.28 | kaldemar | do another check then. |
| 11:17.40 | v0lZy | yeah but im a bit lost on the logic of it |
| 11:17.53 | v0lZy | is chain forwarding common or not? |
| 11:18.26 | v0lZy | I mean... people might not know whos in and whos out and their stnadard workflow might involve forwarding to somone whos forwarding as well because they are both on vacation or something. |
| 11:18.43 | v0lZy | I suppose i should be storing this in the database and check the chain... |
| 11:20.15 | v0lZy | but this is hard... a person could arbitraryly transfer to someone and create a loop. |
| 11:24.28 | dym | "> Channel Local/00011111@faxout-4a88;1 was answered." Does this mean "Local/00011111@faxout-4a88" is the channelid of that call? |
| 11:30.04 | din3sh | yawns |
| 11:40.10 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 11:41.07 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 11:42.07 | Ice_Strike | Is Asterisk 1.6 no longer updated? |
| 11:42.48 | Ice_Strike | I don't see it on the download page |
| 11:44.48 | Ice_Strike | Hmmm should I install Asterisk 1.8 or 10? Which one provide better performance and number of calls? |
| 11:45.09 | kaldemar | Ice_Strike: the last 1.6.X branch got EOL in april: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 11:45.30 | kaldemar | 10 should be better by performance. |
| 11:45.51 | *** join/#asterisk DynamicFail (~Administr@198.99.129.157) |
| 11:46.23 | Ice_Strike | kaldemar Ahhh I see, no wonger I have Asterisk 1.6 installed for while. |
| 11:46.41 | DynamicFail | Does Dundi make it so you essentially define your local extensions one one asterisk server and then it publishes those extensions to other asterisk servers to create automatic trunks based on what is currently active? |
| 11:46.53 | *** join/#asterisk tamiel (~tamiel@c-69-181-254-90.hsd1.ca.comcast.net) |
| 11:47.57 | kaldemar | DynamicFail: something like that. except DUNDi does not publish anything, it just answers to requests. the protocol is reactive. |
| 11:49.06 | DynamicFail | kaldemar, I recently installed freepbx and I currently have a test setup with two asterisk/freepbx servers, each with 3 polycom 335 registered to their respective server. Is there a way to configure DUNDi to test it through freepbx? |
| 11:49.45 | kaldemar | i don't know. ask in #freepbx. |
| 11:50.19 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:54.58 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 11:57.45 | rolandow | YESSSS finally solved my one way audio problem with an Experia box!! :) |
| 11:57.46 | rolandow | happy |
| 11:57.54 | rolandow | and it was NOT my fault :) |
| 12:00.32 | eXcAliBuR | looks at his digium card in his hand and wonders why it isn't working... I have the phone line plugged into it :[ |
| 12:01.02 | *** join/#asterisk Bullmoose (~Bullmoose@75-174-79-252.bois.qwest.net) |
| 12:01.22 | rolandow | you have enough power in your hand? |
| 12:01.30 | eXcAliBuR | well yuh |
| 12:01.56 | rolandow | did you try your other hand? |
| 12:02.20 | eXcAliBuR | if i lick the shiny pins at the bottom, should i get a shock like a 9v battery? |
| 12:02.50 | rolandow | DON'T lick it!! you may get killed! |
| 12:03.10 | rolandow | whatever you do, don't ever lick |
| 12:03.19 | rolandow | well .. at least not with hardware |
| 12:03.25 | eXcAliBuR | :< |
| 12:03.42 | rolandow | not that i'm an expert, but i DO know you shouldn't just start licking things if stuff doesn't work. |
| 12:03.53 | eXcAliBuR | i'll just order a new carrd |
| 12:03.59 | eXcAliBuR | this one must be bust |
| 12:04.17 | rolandow | so you already licked it? |
| 12:04.23 | eXcAliBuR | :[ maybe |
| 12:04.50 | rolandow | so did you get a shock? |
| 12:04.52 | rolandow | curious |
| 12:04.54 | eXcAliBuR | no |
| 12:05.08 | rolandow | oh |
| 12:05.47 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-gkrtmwtiujwqywbi) |
| 12:05.47 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:05.59 | eXcAliBuR | :P i'm just joking, trying to entertain |
| 12:06.13 | rolandow | you're doing a good job |
| 12:06.16 | eXcAliBuR | all done... |
| 12:08.03 | din3sh | licking cards? :o |
| 12:08.20 | dym | digium hardware tastes of raspberry |
| 12:08.22 | Ice_Strike | Do you recommend I install asterisk from source or yum/rpm ? |
| 12:08.33 | dym | Ice_Strike: source |
| 12:08.47 | Ice_Strike | I want to install Asterisk 10 on centos 6.x |
| 12:09.01 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 12:09.32 | din3sh | source |
| 12:09.50 | din3sh | raspberry coated cards |
| 12:10.09 | Ice_Strike | im still waiting for pi raspberry |
| 12:10.13 | rolandow | din3sh: are they backwards compatible with blackberry? |
| 12:10.13 | Ice_Strike | what do you use it for? |
| 12:10.33 | dym | I've got mine (: |
| 12:10.38 | rolandow | how long are you waiting? i had to wait about two weeks |
| 12:10.45 | dym | Building an octocopter atm - pi based |
| 12:10.52 | dym | its "in the oven" if you want |
| 12:11.04 | din3sh | confused! |
| 12:11.30 | dym | can i catch the hangupcause of a call within the extension, or do i have to use h ? |
| 12:11.33 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 12:11.33 | rolandow | what's an octocopter .. like helicopter with extra .. ehm .. wings? |
| 12:11.43 | dym | rolandow: 8 rotors |
| 12:11.53 | rolandow | rotors or blades? |
| 12:12.01 | dym | rotorblades |
| 12:12.02 | rolandow | one rotor has 4 blades with a helicopter, right? |
| 12:12.04 | rolandow | ok |
| 12:12.05 | rolandow | :) |
| 12:12.17 | dym | blade has a blade |
| 12:12.29 | rolandow | are you developing the software yourself as well? |
| 12:12.34 | rolandow | to control the octocopter? |
| 12:12.39 | rolandow | to steer it |
| 12:12.40 | dym | a friend of mine is |
| 12:12.45 | rolandow | nice |
| 12:14.38 | dym | we're planning to construct it with a maneuverable undermount for a 5D to do aerial shoots |
| 12:15.05 | rolandow | so it has to be a rather big octocopter then to be able to carry that? |
| 12:16.59 | din3sh | you guys gonna use it in Avatar 2? |
| 12:18.01 | dym | rolandow: 10" blades |
| 12:18.22 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:18.28 | dym | din3sh: nah, but transformers maybe |
| 12:20.09 | din3sh | :D |
| 12:20.19 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:20.22 | din3sh | how will asterisk control that? |
| 12:22.52 | thecardsmith | if I had Set(DB(foo/bar/1) = "target_value") and I have another db entry say, Set(DB(foo/index) = 1) |
| 12:23.11 | thecardsmith | and I want to retrieve DB(foo/bar/N) by the value of foo/index, somehow I can't seem to do: |
| 12:23.28 | thecardsmith | Noop(${DB(foo/bar/{DB(foo/index)})}) |
| 12:23.39 | thecardsmith | is there a better way? |
| 12:24.14 | thecardsmith | ack I forgot the $ sign in the secon deref there. but, it's there in my dialplan |
| 12:24.32 | thecardsmith | Noop(${DB(foo/bar/${DB(foo/index)})}) |
| 12:25.11 | Ice_Strike | What port should I open for asterisk and it will use SIP connection? |
| 12:25.40 | thecardsmith | Ice_Strike: you're going to want 5060, AND 10,000-20,000 -- you can change the range of 10,000-20,000 to something else in rtp.conf |
| 12:26.07 | Ice_Strike | 10,000-20,000 ? woooh! |
| 12:26.24 | Ice_Strike | what the recommended port for rtp ? |
| 12:26.32 | Ice_Strike | Will be using 200 concurrent channels |
| 12:26.39 | thecardsmith | i think that means you need 400 |
| 12:26.53 | Roelt | and UDP port 4569 for IAX |
| 12:26.54 | [TK]D-Fender | thecardsmith, your set is bad |
| 12:27.01 | kaldemar | thecardsmith: using proper syntax is a bit better. |
| 12:27.27 | [TK]D-Fender | <thecardsmith> if I had Set(DB(foo/bar/1) = "target_value") and I have another db entry say, Set(DB(foo/index) = 1) <- both bad |
| 12:27.42 | thecardsmith | really? |
| 12:28.01 | thecardsmith | the whitespace? |
| 12:28.06 | kaldemar | spaces and quotes are actually characters. |
| 12:28.08 | [TK]D-Fender | thecardsmith, First you never ever put quotes. Every quote you put there is literal content. Second so are the SPACES you have on either side of the = |
| 12:28.29 | thecardsmith | gotchya, ok, yeah the verbose output gave a warning too |
| 12:28.34 | thecardsmith | thanks for that pointer |
| 12:28.36 | [TK]D-Fender | thecardsmith, So you made a key with a SPACE in it. Completely broken |
| 12:28.42 | thecardsmith | gets a coffee, anyone want one? |
| 12:31.28 | Roelt | cooffeeeeeeeeeeeeeeeeeeeee |
| 12:32.13 | Ice_Strike | iptables -A INPUT -p udp -m udp --dport 5060 -j ACCEPT |
| 12:32.16 | Ice_Strike | do I need tcp? |
| 12:32.57 | Roelt | for some phones it's better if they are behind nat |
| 12:32.58 | schmidts | Ice_Strike do you want to use tcp? |
| 12:33.51 | Ice_Strike | what the default? |
| 12:33.52 | schmidts | Roelt tcp doesnt help for a nat setup, it only helps for dropped packets and too large sip packets but not for nat |
| 12:33.59 | schmidts | ice_strike udp ;) |
| 12:34.13 | Ice_Strike | so I just use udp |
| 12:34.57 | Ice_Strike | So I need: |
| 12:34.58 | Ice_Strike | <PROTECTED> |
| 12:35.05 | Ice_Strike | iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT |
| 12:35.07 | Ice_Strike | that it? |
| 12:37.40 | din3sh | why do we have to use such a high range (10000 - 20000)? |
| 12:38.05 | *** join/#asterisk darthanubis (~anubis@unaffiliated/darthanubis) |
| 12:38.29 | Ice_Strike | its default |
| 12:39.13 | *** join/#asterisk slidesinger-lt (~jtatum@c-174-57-5-70.hsd1.nj.comcast.net) |
| 12:39.16 | Ice_Strike | Why it wont start |
| 12:39.19 | Ice_Strike | /etc/init.d/asterisk start |
| 12:39.30 | Ice_Strike | nable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 12:40.16 | Ice_Strike | see http://pastebin.com/RtNsXW2W |
| 12:41.12 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 12:41.46 | *** part/#asterisk Bullmoose (~Bullmoose@75-174-79-252.bois.qwest.net) |
| 12:42.36 | kaldemar | Ice_Strike: "asterisk -vvvc" may tell you. |
| 12:43.21 | Ice_Strike | kaldemar I can see it say "Asterisk Ready." |
| 12:45.06 | kaldemar | Ice_Strike: then you're probably running the init script as the wrong user. maybe you should use sudo or run it as root. |
| 12:45.41 | Ice_Strike | kaldemar I got it to work.. I just typed: asterisk |
| 12:45.46 | Ice_Strike | then it ran in the background |
| 12:45.55 | Ice_Strike | then I did: asterisk -r |
| 12:46.01 | Ice_Strike | I can see CLI now |
| 12:46.19 | bulkorok | Ice_strike: it seems like there is no init script installed |
| 12:46.22 | Ice_Strike | kaldemar I am running as root |
| 12:46.34 | Ice_Strike | bulkorok How to check that on centos? |
| 12:46.37 | bulkorok | ah... kaldemar told sth like that... |
| 12:46.46 | bulkorok | ew... /etc/init.d/ !? |
| 12:46.57 | Ice_Strike | oh you referring to that |
| 12:47.49 | bulkorok | I suppose that... I used debian or ubuntu in the past for asterisk... never tried it with centos... |
| 12:48.04 | Ice_Strike | /etc/init.d/asterisk does exist |
| 12:48.12 | [TK]D-Fender | Ice_Strike, di you install the init script for it? |
| 12:48.58 | Ice_Strike | No? |
| 12:49.09 | thecardsmith | [TK]D-Fender & kaldemar -- worked perfectly after your pointers. If you're ever in Burlington, VT -- message me to take advantage of free beer :) thanks guys |
| 12:49.51 | *** join/#asterisk bchia (~Adium@nat/digium/x-yuortdfqphtwduim) |
| 12:50.08 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 12:50.08 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 12:50.21 | [TK]D-Fender | Ice_Strike, that shouldn't be a question.... |
| 12:50.30 | Ice_Strike | [TK]D-Fender Oh I did do: make config |
| 12:50.34 | Ice_Strike | https://wiki.asterisk.org/wiki/display/AST/Installing+Initialization+Scripts |
| 12:50.48 | Ice_Strike | This work |
| 12:50.51 | Ice_Strike | [root@rapid asterisk-10.7.0]# /etc/init.d/asterisk stop |
| 12:50.51 | Ice_Strike | Stopping safe_asterisk: [ OK ] |
| 12:50.51 | Ice_Strike | Shutting down asterisk: [ OK ] |
| 12:51.17 | Ice_Strike | but not when I start asterisk.... (/etc/init.d/asterisk start) |
| 12:51.26 | [TK]D-Fender | on centos : service asterisk start |
| 12:51.28 | [TK]D-Fender | ^^ |
| 12:51.30 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 12:52.06 | Ice_Strike | [root@rapid asterisk-10.7.0]# service asterisk start |
| 12:52.07 | Ice_Strike | Starting asterisk: |
| 12:52.21 | Ice_Strike | then I did: asterisk -r |
| 12:52.27 | Ice_Strike | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 12:52.48 | Ice_Strike | If I just type: asterisk and then it will run in the background |
| 12:54.38 | [TK]D-Fender | it shows nothing? |
| 12:55.01 | [TK]D-Fender | You said you can't find the INIT script at all.. |
| 12:55.12 | [TK]D-Fender | go set that up again |
| 12:55.30 | Ice_Strike | I did |
| 12:57.29 | mazpe | Ice_Strike: i had a similar problem. |
| 12:57.35 | mazpe | yet asterisk -cvvvvv would start it |
| 12:57.56 | [TK]D-Fender | Go look at your asterisk.conf make sure the paths are all right |
| 12:58.24 | [TK]D-Fender | verify the user the configs are owned by, who * is being run as, etc... |
| 12:58.43 | Ice_Strike | ok sur |
| 12:58.44 | Ice_Strike | sure |
| 12:58.57 | mazpe | cant tell you what it was... as a few reboots later it went away.. and this was happening on different new boxes at the same time |
| 13:03.37 | Ice_Strike | what is this |
| 13:03.38 | Ice_Strike | [root@rapid asterisk-10.7.0]# sudo /etc/init.d/asterisk status |
| 13:03.39 | Ice_Strike | asterisk (pid 19195 19069 18999 18928 18857 18786 18709 18636 18565) is running... |
| 13:04.33 | Ice_Strike | ps aux | grep asterisk |
| 13:04.36 | Ice_Strike | I see a few running lol |
| 13:06.08 | [TK]D-Fender | Ice_Strike, None of that answers any of the permissions question I asked about. |
| 13:06.15 | [TK]D-Fender | Or even basic file locations. |
| 13:06.25 | [TK]D-Fender | moves on to more productive matters. |
| 13:06.49 | Ice_Strike | [TK]D-Fender Dont be rude please, I am looking into it. |
| 13:09.52 | *** join/#asterisk pavlz (~root@unaffiliated/pavlz) |
| 13:09.55 | pavlz | hi |
| 13:10.12 | pavlz | it can seems strange as request for you |
| 13:10.53 | pavlz | i got a linksys wrt54gl on which will be installed OpenWRT |
| 13:11.33 | pavlz | between the RS-232 i would to connect a circuit that must distribute the water to the flowers |
| 13:12.38 | pavlz | i would that ASTERISK at any hours of the day and night with the keyboard of a phone, with sms, e-mail, provide to activate the circuit to distribute the water |
| 13:12.48 | pavlz | is it possible to do it ? |
| 13:14.54 | *** join/#asterisk krdian (krdian@killer.radom.net) |
| 13:15.22 | [TK]D-Fender | pavlz, You can have * call scripts on your system that do that action |
| 13:15.32 | [TK]D-Fender | pavlz, "core show application system" |
| 13:15.42 | [TK]D-Fender | pavlz, "core show function SHELL" |
| 13:17.02 | thecardsmith | pavlz: i'm doing a similar thing with a raspberrypi. i have * on a pi, and I have the pi speaking rs232 to a micro (which then is the brains for my circuit, and the raspi/linux/asterisk/programming language holds my "business logic")... then I just call AGI scripts that do the talking over serial. |
| 13:17.51 | Katty | hello my asterisk does not work at all how to fix?? answer plz. |
| 13:18.14 | thecardsmith | Katty: defenestrate! immediately!!!!1!!!one! |
| 13:18.39 | dym | im trying something wild. i need to cut the last 5 digits from an extension and put them into a seperate variable. how would i go on about doing that? |
| 13:19.05 | polomolo747 | tzafrir_laptop, no, i compiled it without sip support, but now it is fixed, i can connect and call but i can't send message under sip. |
| 13:19.05 | thecardsmith | dym: Set(var=${EXTEN:-5}) |
| 13:19.29 | dym | lol |
| 13:19.35 | dym | thecardsmith: so obvious, but yet so far :D |
| 13:19.40 | dym | cheers buddy |
| 13:19.45 | Katty | thecardsmith: you're using big words |
| 13:19.47 | Katty | thecardsmith: it's 8 am |
| 13:19.49 | polomolo747 | i get the following errors: http://pastebin.com/CCgNnN6D |
| 13:19.52 | Katty | thecardsmith: little words. little |
| 13:19.53 | thecardsmith | dym: hahah, i hear ya :) thanks / no prob |
| 13:20.17 | thecardsmith | hahaha, i'm too many coffees deep, Katty!! |
| 13:20.18 | *** join/#asterisk _Corey_ (~chatzilla@pool-72-78-178-17.phlapa.fios.verizon.net) |
| 13:20.29 | Katty | i want energy |
| 13:20.31 | Katty | poutyface |
| 13:20.46 | Ice_Strike | [TK]D-Fender All the path seem to be fine |
| 13:23.03 | pavlz | mmm, very interesting |
| 13:23.31 | pavlz | at the moment i am giving the water to flowers and to watermelons |
| 13:23.57 | pavlz | which type of circuit are you using for Asterisk with RS-232 ? |
| 13:25.35 | pavlz | i saved your messages with nano, so now i have in the file *.flowers |
| 13:26.01 | *** join/#asterisk Jinxed- (~Administr@147.177.63.14) |
| 13:26.17 | Jinxed- | anyone know why the asterisk/freepbx install doesn't have a dundi.conf file? |
| 13:26.20 | pavlz | do you work in *.py ? i never used *.py |
| 13:26.54 | pavlz | what is AGI ? |
| 13:27.08 | dym | thecardsmith: say i had an extension called 01111.001 - would it be possible to cut everything after the . and put it into a variable, regardless of how many digits? |
| 13:27.30 | WIMPy | core show function CUT |
| 13:27.49 | dym | checks |
| 13:28.26 | Ice_Strike | it say asterisk is running |
| 13:28.27 | Ice_Strike | [root@rapid ~]# /etc/init.d/asterisk status |
| 13:28.27 | Ice_Strike | asterisk (pid 1821) is running... |
| 13:28.41 | Ice_Strike | I can see when I do ps aux | grep asterisk |
| 13:29.00 | Ice_Strike | but it wont let me go to cli when I do asterisk -r |
| 13:29.14 | Ice_Strike | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 13:30.05 | Ice_Strike | I can see asterisk.pid is created but not asterisk.ctl |
| 13:30.14 | *** join/#asterisk serafie (~erin@nat/digium/x-djukywzqgojtdjht) |
| 13:30.57 | [TK]D-Fender | FIX THAT |
| 13:33.09 | Ice_Strike | how? |
| 13:33.27 | pavlz | AGI it's a script, ok, but anyway you wrote the source code, |
| 13:33.40 | pavlz | which are the term of license ? |
| 13:33.47 | pavlz | s |
| 13:34.42 | [TK]D-Fender | Ice_Strike, asterisk.conf <--- and your init script |
| 13:35.00 | [TK]D-Fender | pavlz, How do you control them on your own right now? |
| 13:36.49 | thecardsmith | pavlz: i can share my work, if you'd like. but, for the most part it's nothing "super new", I'm just using a perl module to talk to /dev/ttyAMA0: http://search.cpan.org/~cook/Device-SerialPort-1.002/SerialPort.pm and then I call AGI(/usr/bin/myPerl.pl) |
| 13:38.16 | pavlz | ok, which is the more flexible circuit to be associated to ASTERISK and in particular to AGI script, to the a similar job as your ? |
| 13:40.05 | pavlz | isn't *.pm PERL ? |
| 13:40.18 | thecardsmith | perl module |
| 13:40.24 | pavlz | correct |
| 13:40.33 | pavlz | good language |
| 13:40.38 | Ice_Strike | [TK]D-Fender I have got it fixed... set SELINUX=disabled in /etc/selinux/config |
| 13:41.04 | *** join/#asterisk UQlev (~chatzilla@46.251.117.25) |
| 13:41.07 | pavlz | amy say that is dirty, and that is for crazy people which want a hard life |
| 13:41.22 | pavlz | many |
| 13:42.03 | [TK]D-Fender | pavlz, You probably don't need an AGI for this... |
| 13:42.10 | thecardsmith | pavlz: have you played with an Arduino before? It's an easy-to-use platform, and good for mocking up ideas, and... you can talk serial to it. It's a good jumping point for using microcontrollers. I know, I'm getting off topic, but... hooking up asterisk to "the physical world" is cool to me :) |
| 13:43.31 | pavlz | no, at the moment |
| 13:43.49 | din3sh | SELINUX is EVIL!!!!!!!!!!! |
| 13:44.03 | pavlz | SELINUX is EVIL ? |
| 13:44.07 | pavlz | why ? |
| 13:44.18 | thecardsmith | it sure is a pain in the ass XD ...selinux |
| 13:45.10 | din3sh | lol |
| 13:45.24 | din3sh | its evil cause it makes you pull your hair off |
| 13:46.19 | pavlz | http://www.nsa.gov/research/selinux |
| 13:47.18 | thecardsmith | pavlz, [TK]D-Fender is probably right, you can probably do something simply like: System(echo "data" > /dev/ttySERIALDEVICE) (where "SERIALDEVICE" is your serial device) |
| 13:49.12 | thecardsmith | I use an AGI script because I wanted to be a little more intimate with the serial communications, but, it doesn't mean you have to |
| 13:49.29 | pavlz | i thank you many of you, for the help |
| 13:49.59 | thecardsmith | no prob, hey pavlz if you want to /msg me your email address, I can send you an email if I document my project in a blog etc etc |
| 13:51.30 | pavlz | it's very interesting, but i am sorry, i have to go to the Chamber of Commerce, Industry, Handycract, Agriculture, |
| 13:51.48 | pavlz | here in rome, italy |
| 13:54.38 | thecardsmith | tell the pope I like his hat |
| 13:56.11 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 13:56.55 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 13:59.36 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 14:00.50 | dym | exten => _X.,n,Set(__FUID1=${CUT(EXTEN,*,1)}) doesnt seem to work. I have an extension thats 23423423*123 and i need to seperate the 23423423 and the 123 into seperate variables. |
| 14:01.30 | dym | ~(this will obviously only set one, but you get the drift) |
| 14:03.53 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:03.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:04.34 | [TK]D-Fender | dym, PB the full call and code |
| 14:12.06 | dym | [TK]D-Fender: http://pastebin.com/eMA39TwJ |
| 14:12.27 | dym | (call is initiated via a callfile) |
| 14:14.37 | *** join/#asterisk PadawanPete (~kvirc@91.208.243.2) |
| 14:15.05 | *** join/#asterisk webguynow (~webguynow@c-24-1-222-204.hsd1.il.comcast.net) |
| 14:15.08 | PadawanPete | Hello, is this the correct channel to perhaps ask for asterisk assistance if you are stuck in a certain area? |
| 14:16.06 | dym | PadawanPete: Shoot your question and find out |
| 14:16.35 | *** join/#asterisk n3hxs (~ed@ann100fw01.answernet.com) |
| 14:17.10 | PadawanPete | I am trying to find out more information on how presence information (online, offline etc) works, since I am battling to get it to work with the softphone I am using. |
| 14:17.37 | PadawanPete | Does anyone know what all the different options are, and what to do if its not working out for you? |
| 14:21.03 | drmessano | Which version of Asterisk are you using? |
| 14:21.05 | kaldemar | dym: CUT(EXTEN,*,-3) <--- "-3"? |
| 14:22.06 | PadawanPete | im using asterisknow just downloaded it yesterday off the website. |
| 14:22.56 | PadawanPete | 3.2.22 according to yum |
| 14:23.42 | drmessano | 3.2.22 is neither an AsteriskNOW version or an Asterisk version |
| 14:23.55 | PadawanPete | oh right maybe I did that wrong |
| 14:24.00 | PadawanPete | let me try again |
| 14:24.07 | [TK]D-Fender | dym, OH, that cut isn't right |
| 14:24.25 | [TK]D-Fender | dym he showed you a var regference way to chop a fixed # fof chars |
| 14:24.33 | [TK]D-Fender | that isn't how you use the CUT function for this however |
| 14:25.00 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:25.15 | PadawanPete | asterisk-core.x86_64 1.8.11-0.cert5.1_centos5 installed |
| 14:25.32 | PadawanPete | Is that better or ...? |
| 14:26.19 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:26.55 | [TK]D-Fender | PadawanPete, Set up your hints. Have phones register. Have phones subscribe to the hints |
| 14:27.24 | Katty | Qwell: ping |
| 14:28.12 | PadawanPete | D-Fender, thank you for your reply, however I am uncertain how I should do this :( or what process I need to follow. I did see someone suggesting the same thing on google, somewhere, but was unable to understand the document :( |
| 14:32.40 | [TK]D-Fender | ? book |
| 14:32.50 | [TK]D-Fender | and look up "hints" on the WIKI |
| 14:32.54 | [TK]D-Fender | very basic |
| 14:33.13 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 14:34.35 | PadawanPete | D-Fender, I will try to do as you ask, but any additional help would be appreciated since I am battling to start. Thank you for your time anyway :) |
| 14:34.58 | drmessano | If you're using AsteriskNOW, hints are generated by the FreePBX dialplan. All you need to do is add them in your softphone |
| 14:35.24 | drmessano | Add 101@host and you're done |
| 14:35.27 | drmessano | For example |
| 14:35.40 | drmessano | I think Jitsi and Xlite are that easy.. Not sure about the rest |
| 14:36.19 | PadawanPete | Interesting, thanks drmessano. Perhaps my softphone of choice does not support this option? |
| 14:36.43 | [TK]D-Fender | PadawanPete, "core show hints" <- pastebin this |
| 14:36.44 | [TK]D-Fender | ? pb |
| 14:36.48 | [TK]D-Fender | ~pb |
| 14:36.48 | 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. |
| 14:36.49 | [TK]D-Fender | ^^^ |
| 14:37.55 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:38.23 | PadawanPete | Thank you for the additional help, the command I have decided to execute is as follows : asterisk -r -x "core show hints" |
| 14:38.39 | PadawanPete | I will paste the output at the site above one moment. |
| 14:39.39 | PadawanPete | Here is the output :) http://pastebin.com/Uv0fYGXi |
| 14:41.33 | *** join/#asterisk navaismo (~navaismo@189.191.94.251) |
| 14:44.53 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 14:47.22 | dym | [TK]D-Fender: How would i go on accomplishing the chopping then? |
| 14:47.36 | dym | i need the bit after * in one var and the bit before * in another |
| 14:54.05 | mazpe | 10:36:49 < [TK]D-Fender> ^^^ |
| 15:01.03 | PadawanPete | Did the output seem correct... ? |
| 15:02.17 | *** join/#asterisk PadawanPete (~kvirc@91.208.243.2) |
| 15:02.30 | PadawanPete | Hello, did anyone respond to the pastebin? |
| 15:03.00 | Qwell | Katty: HRM? |
| 15:05.09 | drmessano | PadawanPete, yes.. looks fine |
| 15:06.32 | PadawanPete | ok thanks |
| 15:07.13 | Katty | Qwell: hai *hug* |
| 15:07.39 | Qwell | ohai |
| 15:09.12 | PadawanPete | OK will try with a different softphone |
| 15:10.21 | drmessano | PadawanPete, Like I said.. The FPBX dialplan does all the work as far as hints. I never had to tweak anything.. Standard setup, then add the extension to my softphone as sip:exten@host or exten@host or sip:exten .. Depends on the phone |
| 15:10.44 | drmessano | So you have a good baseline.. Just find a softphone and play with the settings |
| 15:13.33 | PadawanPete | I am testing instead with jitsi and it worked without altering anything. |
| 15:13.48 | PadawanPete | I am assuming that the one I tried doesnt support it. |
| 15:15.17 | drmessano | Could be |
| 15:16.06 | [TK]D-Fender | dym, CUT will do it, but you combined his instructions which were for a DIFFERENT method |
| 15:16.13 | [TK]D-Fender | dym, "core show function CUT" <----------- |
| 15:16.20 | [TK]D-Fender | Read CUT's instructions |
| 15:16.26 | [TK]D-Fender | -3 is not what you're looking for |
| 15:17.07 | [TK]D-Fender | PadawanPete, well you have hints.. just no phone has asked to WATCH them yet |
| 15:17.30 | *** join/#asterisk Galen (~Galen@rrcs-24-43-20-118.west.biz.rr.com) |
| 15:22.33 | PadawanPete | yes probably I was testing with linphone |
| 15:25.11 | Katty | you should test with a banana phone instead. |
| 15:25.37 | PadawanPete | Thank you for your input, however I may decline your offer. |
| 15:26.26 | _Corey_ | So, I've had this happen a few times on a couple 1.8.11-cert5 machines... Asterisk will stop processing SIP calls randomly--actually turning up debugging shows it not seeing the SIP messages. They're visible coming in (confirmed on the OS level). Restarting Asterisk fixes. Anyone seen this? |
| 15:27.00 | Qwell | _Corey_: sounds like a (partial) deadlock, maybe |
| 15:27.42 | _Corey_ | hmm, how would I confirm that? |
| 15:27.55 | Katty | i totally read that as particle deadlock |
| 15:27.58 | Katty | and got all excited for a moment |
| 15:28.42 | Qwell | Katty: Only if you use Asterisk in Switzerland. |
| 15:29.09 | Katty | sets up a hybrid particle swarm algorithm |
| 15:29.16 | Katty | LET"S FLY |
| 15:29.34 | Qwell | _Corey_: build with thread debugging enabled, and grab a 'core show locks' if it happens again |
| 15:29.51 | PadawanPete | ok thanks everyone bye bye :) |
| 15:30.32 | _Corey_ | Qwell: OK, thx... |
| 15:33.14 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 15:33.16 | mazpe | any preference of patton vs audio codes fxo gateways? |
| 15:38.26 | *** join/#asterisk danfromuk (~IceChat77@2.27.27.15) |
| 15:52.25 | *** join/#asterisk fexilal (~felixal@unaffiliated/felixal) |
| 15:52.30 | fexilal | Hi |
| 16:08.15 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:25.26 | *** join/#asterisk din3sh (~din3sh@41.136.87.52) |
| 16:25.47 | din3sh | how to implement email to fax ? |
| 16:28.15 | [TK]D-Fender | go look at your fax apps. |
| 16:28.21 | [TK]D-Fender | use the send one you have installed. |
| 16:28.29 | [TK]D-Fender | Code the rest yourself ... |
| 16:32.45 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 16:33.39 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:38.53 | *** join/#asterisk lorsungcu (~lorsungcu@65.103.31.36) |
| 16:39.23 | lorsungcu | anyone have experience with swift()? |
| 16:40.59 | lorsungcu | i need to match against 1-5 digits, but it keeps grabbing just the first digit entered |
| 16:42.27 | lorsungcu | here's the command |
| 16:42.31 | lorsungcu | Swift("<prosody rate=${swiftRate} pitch=${swiftPitch} volume=${swiftVolume}>${swiftWelcomeThankYou}</prosody>",${swiftWait},5)) |
| 16:48.19 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:48.37 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-193.mobile.uci.edu) |
| 17:01.10 | din3sh | what would be the hardware requirements (server) for 600 SIP phones, around 300 concurrent calls, alaw? |
| 17:08.53 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 17:10.09 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:11.30 | *** join/#asterisk danfromuk (~IceChat77@2.27.27.15) |
| 17:12.03 | *** join/#asterisk rneese (~Richard@97.100.95.108) |
| 17:12.28 | rneese | so other then the freepbx any other new guis for asterisk ? |
| 17:12.37 | rneese | that anyone is using |
| 17:12.38 | [TK]D-Fender | FreePBX is not "new" |
| 17:12.43 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 17:13.07 | [TK]D-Fender | Nor is AsteriskGUI (practically dead), or several others |
| 17:13.25 | rneese | yeah I kow they killed the digium gui |
| 17:13.30 | rneese | why who knows |
| 17:13.39 | rneese | it was a nice clean little gui |
| 17:13.42 | [TK]D-Fender | lack of money to be made by it |
| 17:13.46 | [TK]D-Fender | They bought out Switchvox. |
| 17:13.53 | [TK]D-Fender | That's where their focus is |
| 17:14.13 | mazpe | I just made my own :) |
| 17:14.23 | rneese | ok |
| 17:14.34 | mazpe | perl+catalyst-framework |
| 17:15.51 | rneese | I figured someone woud pickup and keep the digium gui going |
| 17:16.16 | rneese | never played with switchvox |
| 17:17.14 | [TK]D-Fender | No-one does. That garden's walls are HIGH |
| 17:17.43 | rneese | lol |
| 17:18.11 | [TK]D-Fender | And AsteriskGUI ... well... it was never properly modular, depended on users.conf, and was too toaster-iffic for me to see it going far |
| 17:20.47 | rneese | so the only other one I know of was a paid gui called thirdlane |
| 17:21.07 | rneese | so what is everyone using ? |
| 17:21.16 | lorsungcu | thirdlane is rough |
| 17:21.18 | lorsungcu | not a fan |
| 17:21.44 | [TK]D-Fender | rneese, What do you NEED? |
| 17:22.00 | rneese | yeah I know alex the developer and he is not open to input much |
| 17:22.14 | lorsungcu | that was my experience, rneese. |
| 17:22.52 | rneese | well I am getting pissed off at the freeswitch group as I have been commiting bug fixes and then they sit in the jira and . and not been commited |
| 17:23.24 | rneese | so I am just looking for something I can toss up on a macjine for use at home |
| 17:23.40 | rneese | but last I knew freepbx was dying off |
| 17:23.50 | lorsungcu | not at all? |
| 17:23.53 | Qwell | rneese: Where did you hear that nonsense? |
| 17:23.55 | rneese | and thridlane not worth the fight |
| 17:24.20 | drmessano | FreePBX is absolutely not dying off lol |
| 17:24.27 | rneese | ok |
| 17:24.35 | rneese | then I can look at it |
| 17:24.43 | lorsungcu | they |
| 17:24.47 | [TK]D-Fender | I'd like to hear where you heard that from... |
| 17:25.04 | lorsungcu | switchvox sales guy |
| 17:25.30 | drmessano | Switchvox sales guy said FreePBX is dying off? |
| 17:25.46 | dym | wonder why :D |
| 17:25.48 | lorsungcu | rofl no |
| 17:25.51 | din3sh | what would be the hardware requirements (server) for 600 SIP phones, around 300 concurrent calls, alaw? |
| 17:25.56 | rneese | back when it was moved to 2600 hz |
| 17:26.08 | [TK]D-Fender | drmessano, no, that was rneesewho said that... |
| 17:26.09 | rneese | adn then became bluebox and bluebox died |
| 17:26.09 | Qwell | It was never "moved" anywhere... |
| 17:26.19 | lorsungcu | ... rneese, i think you are confusing freeswitch/freepbx |
| 17:26.44 | rneese | no bluebox was freepbx that did both astrisk and freeswitch |
| 17:26.55 | [TK]D-Fender | Bluebox was a SPIN-OFF project and mainline FreePBX continued on just fine |
| 17:26.59 | rneese | I guess it was just s branch of freepbx |
| 17:27.04 | rneese | ok |
| 17:27.11 | drmessano | meese: The story is simple. There was an attempt to roll a FreePBX 3.x as a total rewrite, and it went a different direction.. so the project was split off and renamed. FreePBX 2.x remainds in active development |
| 17:27.32 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 17:28.38 | rneese | then I will have to get a list of deps and attempt to make a bsd port fo it so I can use it on bsd |
| 17:28.52 | lorsungcu | just install the distro |
| 17:28.55 | rneese | till then I will install deb and build a box |
| 17:29.02 | lorsungcu | you will be much happier |
| 17:29.16 | Qwell | Nobody uses BSD. |
| 17:29.34 | drmessano | Qwell, there's this one guy.. but I think he died |
| 17:29.42 | Qwell | Fine, move my 's' over a few characters. |
| 17:29.55 | rneese | yes they do |
| 17:29.57 | rneese | I do |
| 17:30.00 | rneese | I live on it |
| 17:30.12 | [TK]D-Fender | rneese, Around here you're "nobody" ;) |
| 17:30.13 | lorsungcu | bsd + debian? |
| 17:30.15 | rneese | the iso is it deb or centos |
| 17:30.22 | lorsungcu | centos |
| 17:30.27 | rneese | grrr |
| 17:30.33 | rneese | not a centos fan |
| 17:30.33 | drmessano | CentOS sucks |
| 17:30.57 | rneese | you should redo it with debian |
| 17:31.12 | drmessano | Debian and Ubuntu installation of FreePBX is pretty simple. |
| 17:31.17 | Qwell | Everybody knows FreePBX doesn't work on Ubuntu. |
| 17:31.22 | drmessano | ROFL |
| 17:31.25 | Qwell | /debian |
| 17:31.33 | drmessano | Thats right Qwell |
| 17:31.59 | drmessano | It's missing ____ and the paths are hardcoded |
| 17:32.01 | drmessano | like / |
| 17:32.06 | drmessano | and /var/www |
| 17:32.09 | drmessano | and and /etc |
| 17:32.12 | drmessano | Bastards |
| 17:32.27 | drmessano | Actually, not at all |
| 17:32.40 | *** join/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210) |
| 17:37.28 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:45.21 | lorsungcu | still need some help with swift, if anyone's around |
| 17:46.47 | *** part/#asterisk rneese (~Richard@97.100.95.108) |
| 17:50.16 | fexilal | anyone know if there exists a template you can give to a customer on how what basic features they want for their extensions? |
| 17:50.30 | fexilal | on what basic* |
| 17:51.55 | lorsungcu | what features are you referring to, fexilal? |
| 17:52.54 | fexilal | things like call groups, outgoing calls, etc. |
| 17:54.13 | fexilal | anything to simply getting the customer's requirements |
| 17:55.26 | mazpe | ok... audio codes support sucks! i'm leaning towards patton for fxos gateway |
| 17:56.00 | lorsungcu | mazpe, ive had the reverse. love me some audiocodes. |
| 17:56.09 | lorsungcu | what sort of trouble are you having? |
| 17:56.20 | mazpe | just getting setup! |
| 17:56.32 | lorsungcu | what model? |
| 17:56.40 | mazpe | obviously, i have no clue what i'm doing.. but the manual its very different from the current models. |
| 17:56.44 | mazpe | mp114 |
| 17:56.53 | mazpe | only 4 ports... shouldnt be this difficult.. lol |
| 17:57.22 | mazpe | I got it to the point, where asterisk is making the call and sending over to the audiocodes.. but its not going out from the audio codes |
| 17:57.38 | *** join/#asterisk cyborg-one (1000@212-178-1-80.broadband.tenet.odessa.ua) |
| 17:57.54 | mazpe | I even see the first green light, on the home page, where it becomes green (RTP Active) |
| 17:58.32 | lorsungcu | hmm. havent used their stuff for fxo, only t1 |
| 17:58.32 | lorsungcu | looking for a manual atm |
| 17:58.49 | mazpe | ive found the one from voipsupply.com |
| 18:03.08 | lorsungcu | see anything in the logs? |
| 18:03.43 | lorsungcu | i seem to recall having some issue with outbound on a mediant1000 |
| 18:03.52 | lorsungcu | it wasnt totally clear what needed to be set |
| 18:04.04 | lorsungcu | but i cant find anything on the mp11x series |
| 18:04.19 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-193.mobile.uci.edu) |
| 18:05.15 | lorsungcu | http://www.voip-info.org/wiki/view/AudioCodes+MediaPack+114+and+118+SIP+Gateway+with+Asterisk |
| 18:05.49 | *** join/#asterisk b0ot (~Administr@147.177.62.46) |
| 18:08.30 | b0ot | In what ways could asterisk provide a decentralized distributed system? I'm looking for keywords so I can go google hunting |
| 18:08.42 | b0ot | I know about DUNDi, but that is mainly just for dn lookup |
| 18:08.54 | seanbright | search for "hackity hack hack hack" |
| 18:09.09 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 18:09.19 | seanbright | i kid |
| 18:09.32 | lorsungcu | what are you trying to do, boot |
| 18:10.32 | rocksfrow | hi, i was just wondering what asterisk's plans were with the 'queue_log' file. i'm just wondering if there are any plans on ditching queue_log or if it's safe to say it's going to stick around for future releases to come. |
| 18:11.16 | rocksfrow | Pretty sure it dates back to 1.4 atleast so I suspect so but just like to confirm. Thanks ahead of time if anybody can. |
| 18:11.41 | Katty | takes a break from OCDing. |
| 18:11.43 | mjordan | rocksfrow: there are no plans to change its functionality or deprecate it at this time. |
| 18:11.59 | rocksfrow | mjordan, awesome, thanks :) |
| 18:12.00 | Katty | home decor is overwhelming:< |
| 18:13.00 | lorsungcu | just give it all away, katty |
| 18:13.08 | lorsungcu | keep a couch and a standing desk |
| 18:13.15 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 18:13.23 | Katty | lorsungcu: no that's the problem, i don't /have/ any |
| 18:13.36 | lorsungcu | you' |
| 18:13.40 | Katty | lorsungcu: after about 3 hours of trying to figure out the bedding situation, i finally decided on wall art >.< |
| 18:13.40 | lorsungcu | re ahead of the game |
| 18:13.57 | b0ot | lorsungcu, My end goal is to have a VoIP solution that isn't dependent on any one server. IE endpoints register with a "cloud" or some layer of abstraction above the servers. There is only one configuration for the entire network. If a server goes down, endpoints are affected assuming they have connection to a different server |
| 18:14.14 | b0ot | if the cloud got split in two both clouds could operate independently |
| 18:14.26 | b0ot | and then when they regained connectivity they could work again |
| 18:15.37 | Katty | lorsungcu: http://ak1.ostkcdn.com/images/products/L13101983.jpg |
| 18:15.58 | lorsungcu | mmm |
| 18:17.16 | Katty | http://www1.macys.com/shop/product/hotel-bedding-frame-lacquer-collection?ID=623550 <- THAT is the great debate. |
| 18:17.21 | Katty | i think it's too busy for the artwork |
| 18:18.13 | Katty | target has something similiar, and a bit more toned down. |
| 18:19.14 | lorsungcu | im more a fan of wool |
| 18:19.31 | Katty | well it's not about the material. |
| 18:19.34 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:19.36 | Katty | it's about the bed over-power the artwork |
| 18:19.43 | Katty | and it's my ocd panties in a wad. |
| 18:19.49 | lorsungcu | http://www.faribaultmill.com/store.html |
| 18:20.08 | Katty | that won't work for my feather blanket. |
| 18:20.11 | Katty | those aren't duvets. |
| 18:20.19 | lorsungcu | bah. |
| 18:20.30 | Katty | but a nice solid grey wool blanket |
| 18:20.34 | Katty | may be the ticket. |
| 18:20.59 | lorsungcu | you wont find anything more comfortable than nice sheets + wool blanket |
| 18:21.14 | Katty | i'm not giving up my feather blanket. |
| 18:21.23 | Katty | we are in a very serious, commited relationship. |
| 18:21.24 | b0ot | Katty, = offsubject master |
| 18:21.56 | *** join/#asterisk gregor3005 (~Benutzern@85-125-11-10.static.xdsl-line.inode.at) |
| 18:22.08 | Katty | but that silver wool blanket would look very nice between sheets and the duvet. |
| 18:22.21 | lorsungcu | you will die of heat |
| 18:22.28 | Katty | winter is coming. |
| 18:22.34 | lorsungcu | i have sheets + wool blanket all year long |
| 18:22.35 | Katty | and my blood is like -50F in winter. |
| 18:22.51 | lorsungcu | whereabouts are you? |
| 18:22.52 | Katty | trust me! i am an anemic living bloodscicle. |
| 18:23.01 | Katty | southern missouri. |
| 18:23.04 | lorsungcu | rofl |
| 18:23.15 | lorsungcu | you will be ok :D |
| 18:23.24 | Katty | of course i will be ok. |
| 18:23.28 | lorsungcu | with only wool |
| 18:23.31 | Katty | i have a feather blanket and a boy. |
| 18:23.36 | Katty | i shan't be cold. |
| 18:23.50 | Katty | unless i get up in the middle of the night to go pee, which always sucks :< |
| 18:24.08 | lorsungcu | http://www.smartwool.com/ |
| 18:25.02 | Katty | HOT DIGGITY knee high socks! |
| 18:25.13 | Katty | bookmark |
| 18:25.17 | lorsungcu | those are some of the nicest socks you can buy.. |
| 18:25.22 | Katty | lorsungcu: <3 |
| 18:25.49 | Katty | oh look, we have a local dealer! |
| 18:26.00 | lorsungcu | get the mountaineering ones if you are cold |
| 18:26.07 | lorsungcu | and you will never be cold again. |
| 18:26.08 | Katty | ok |
| 18:26.14 | Katty | makes note |
| 18:26.25 | lorsungcu | but you may need to buy new shoes as they are like 1/2" thick |
| 18:26.55 | lorsungcu | http://www.smartwool.com/womens/mountaineering-extra-heavy-crew-1189.html |
| 18:27.14 | lorsungcu | i have two pair, they've lasted me ~6 years |
| 18:27.19 | lorsungcu | no holes |
| 18:28.51 | Katty | those look nice, and yes i'd likely have to get bigger shoes. |
| 18:29.42 | Katty | i wear t hose minimialist type running shoes |
| 18:29.58 | Katty | except in winter. i totally sport those ugg type boots |
| 18:29.58 | lorsungcu | ah |
| 18:30.13 | lorsungcu | icebreaker is also very good |
| 18:30.24 | lorsungcu | but usually more expensive |
| 18:30.35 | jpsharp | I live in Florida and people wear uggs year around. |
| 18:30.48 | Katty | jpsharp: those /actual/ uggs are redonkulious |
| 18:30.53 | Katty | they're like house shoes with a sole. meh |
| 18:31.05 | Katty | i get those fur lined lace up ones |
| 18:31.21 | Katty | knee high snow boots? |
| 18:31.25 | jpsharp | I'll stick with my Teva sandals. |
| 18:31.27 | Katty | idk what they're called |
| 18:31.39 | Katty | meh teva sandles |
| 18:31.49 | Katty | don't care for those, but some people swear by them |
| 18:32.10 | jpsharp | Comfy as hell. I can walk 500 miles in them. And then 500 more. |
| 18:32.29 | Katty | ^_____^ |
| 18:32.34 | lorsungcu | yeah i have chacos |
| 18:32.40 | lorsungcu | pretty much awesome |
| 18:32.53 | Katty | meh |
| 18:32.59 | Katty | just not my thing, i spose |
| 18:33.25 | Katty | flip flops, platform pumps, minimalist sneakers, and snow boots. |
| 18:33.28 | Katty | that's all i own >.< |
| 18:33.56 | jpsharp | Not much of a call for snow boots down here. |
| 18:33.57 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 18:34.03 | Katty | yeah. |
| 18:34.09 | Katty | you need like...hurrican slosh boots. |
| 18:34.23 | Katty | golashes! |
| 18:34.51 | lorsungcu | im in minnesota. usually just wear really nice hiking boots all year long |
| 18:34.58 | lorsungcu | + wool socks :p |
| 18:35.19 | Katty | well that's cause you have ice storms 350 days of the year |
| 18:35.21 | Katty | :P |
| 18:35.25 | lorsungcu | psh |
| 18:35.41 | lorsungcu | its like 95 here or something |
| 18:35.52 | Katty | with a chance of ice storms. |
| 18:35.52 | lorsungcu | i just like routine :D |
| 18:36.10 | Katty | routine can be good. |
| 18:36.52 | Katty | jpsharp: i have that song stuck in my head now. dagnabit :< |
| 18:39.55 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:39.55 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 18:41.34 | lorsungcu | still looking for some help with swift(), anyone have experience? |
| 18:41.55 | michael-i | In Asterisk 10+, if I have defined custom fields in my CDR and those columns are present in my postgres table, should the values automatically be stored by cdr_pgsql? |
| 18:42.44 | navaismo | where can I get information about creating my own rpm's of asterisk, its like cross-compile? |
| 18:42.45 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.37) |
| 18:43.06 | jpsharp | Katty: My work here is done :) |
| 18:45.04 | Hive | hey mjordan, you around? |
| 18:46.10 | Hive | learn from my mistakes; you have to Answer() before Queue() in order to hear music on hold and periodic announcements... that is all |
| 18:46.36 | *** join/#asterisk michael-i_ (~michael-i@204.11.231.178.static.etheric.net) |
| 18:46.45 | lorsungcu | is there any harm in always answer()'ing? |
| 18:47.35 | Hive | not that I can think of, but dont take my word as law |
| 18:47.40 | Hive | i'm (clearly) still learning a lot haha |
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| 18:48.56 | *** join/#asterisk gtTuna (Tuna@d118-75-91-174.try.wideopenwest.com) |
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| 18:53.09 | *** join/#asterisk jkroon (~jkroon@dsl-244-38-190.telkomadsl.co.za) |
| 18:54.28 | jkroon | hi all, when doing Dial(*SIP*/peer/number), if the response sequence I get from my upstream (I wanted to puke when I saw this) is 100 Trying, 180 Ringing (WTF?!?), 404 Not Found, what should the HANGUPCAUSE variable be set to? |
| 18:56.13 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 18:57.03 | gtTuna | Can anyone help me understand why I'm not getting to my 'h' extensions after a call has ended http://pastebin.com/xGZk1dcg ?? |
| 18:57.23 | gtTuna | astersik 1.6.2 |
| 18:57.35 | [TK]D-Fender | gtTuna, because you tried using "h" as a PRIORITY |
| 18:57.41 | lorsungcu | jkroon, you play sc2? |
| 18:57.52 | [TK]D-Fender | gtTuna, "h" is an Asterisk Standard EXTENSION |
| 18:58.11 | jkroon | lorsungcu, no |
| 18:58.15 | gtTuna | can I do same => h,1,NoOp("bllaaahhhh") |
| 18:58.16 | gtTuna | then? |
| 18:58.30 | [TK]D-Fender | no, because same = EXTENSION |
| 18:58.36 | [TK]D-Fender | "h" is ANOTHER extension. |
| 18:58.39 | gtTuna | oh |
| 18:58.41 | [TK]D-Fender | It is not related to any other |
| 18:59.59 | gtTuna | hmm, well what I want to do is grab a recorded file after I call StopMixMonitor() |
| 19:01.49 | [TK]D-Fender | So go do it |
| 19:03.17 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:03.17 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:03.24 | gtTuna | trying...seems like anything anything after my Dial() doesn't get called |
| 19:03.51 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:03.51 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 19:04.17 | [TK]D-Fender | gtTuna, I don't see sanely updated configs and CLI output to match.... |
| 19:07.21 | jkroon | ok, in channels/chan_sip.c, around line 20296 (may vary from version to version) the hangup in response to a 404 (the queued hangup) sends a AST_CAUSE_CONGESTION - is this correct? Should that not be AST_CAUSE_UNALLOCATED? |
| 19:08.25 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 19:11.29 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 19:16.09 | gtTuna | [TK]D-Fender, http://pastebin.com/jXsuBQHE |
| 19:18.30 | Nivex | I had my toll-free numbers routed out through sipphone.com way back in the day. Is there a reasonably trustworthy place to send that traffic nowadays? |
| 19:19.30 | [TK]D-Fender | gtTuna, and now I see no "h" exten at all.... |
| 19:19.38 | gtTuna | got rid of it |
| 19:19.51 | [TK]D-Fender | gtTuna, Well isn't that just counter-productive............. |
| 19:20.05 | gtTuna | ... |
| 19:20.05 | *** join/#asterisk nightowl82 (~ameyer@c-67-171-191-178.hsd1.or.comcast.net) |
| 19:20.49 | [TK]D-Fender | gtTuna, Dial gets answered. Someone hangs up. No "h" to fall on. The End. |
| 19:21.02 | [TK]D-Fender | Dialplan doesn't jsut go on after a Dial gets answered. |
| 19:21.15 | [TK]D-Fender | that's what "h" is for. |
| 19:21.20 | [TK]D-Fender | DO IT |
| 19:21.40 | lorsungcu | DOOOOO IT |
| 19:23.31 | generalhan | is there a way to have hint updates not show up in the CLI ... without losing the verbosity of other events i need to see? |
| 19:24.22 | nightowl82 | My sip client registers correctly but when I make a call, it shows that Asterisk is trying to route the call through an invalid ip address. My sip client isn't on the same network. |
| 19:25.15 | generalhan | with 10 expansion modules showing hints for every extension in the company, it really starts to clutter up the CLI to see everytime someone's extension becomes idle/unavailable. but i dont want to lower the verbosity and miss out on things that i actually want to see |
| 19:27.36 | navaismo | when an agi has all permission(777) and owned by the same user running asterisk, but the asterisk cli says: Failed to execute '/var/lib/asterisk/agi-bin/test.agi': Permission denied |
| 19:27.41 | navaismo | what is the next step |
| 19:27.45 | [TK]D-Fender | generalhan, Nope |
| 19:28.08 | generalhan | [TK]D-Fender: i was affraid of that. thanks for the info ! |
| 19:28.58 | generalhan | have to write a live parsing tool to take those out and show me everything else ! lol |
| 19:29.31 | jpsharp | Nivex: I route my TF calls to tollfree.alcazarnetworks.com |
| 19:29.38 | *** join/#asterisk fling (~fling@fsf/member/fling) |
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| 19:31.27 | jpsharp | Nivex: I use them cause they support TLS + SRTP |
| 19:36.31 | Nivex | jpsharp: cool, I'll give it a try. thanks. |
| 19:36.57 | *** part/#asterisk gregor3005 (~Benutzern@85-125-11-10.static.xdsl-line.inode.at) |
| 19:38.40 | *** part/#asterisk nightowl82 (~ameyer@c-67-171-191-178.hsd1.or.comcast.net) |
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| 19:43.49 | nightowl82 | My sip client is connecting from outside the network and when I try to call a test extension, it hangs up without error messages. The call seems to be being routed through an IP address that I didn't specify. What can I do? |
| 19:44.59 | nightowl82 | Instead of being routed back out to the ip address of my sip client, it is being routed to a local ip address that has no host associated with it. |
| 19:45.02 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 19:45.02 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 19:51.01 | navaismo | anyone? |
| 19:51.08 | navaismo | this drive crazy |
| 19:51.17 | navaismo | drive me* |
| 19:53.44 | lorsungcu_ | you have no issue running the script from outside asterisk? |
| 19:54.31 | navaismo | lorsungcu, nope |
| 19:57.22 | Qwell | What is the first line of the script? |
| 19:59.50 | navaismo | Qwell, this --> #!/usr/bin/php -q |
| 20:03.30 | navaismo | here are more information: http://pastebin.com/6bRHY4EK |
| 20:03.41 | navaismo | script, dialplan and the output in the cli |
| 20:08.14 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 20:10.22 | lorsungcu_ | dammit i want a raspberry pi. |
| 20:12.17 | lorsungcu_ | permissions for phpagi/phpagi.php correct? |
| 20:12.51 | lorsungcu_ | or, are you able to run other phpagi scripts? |
| 20:14.35 | drmessano | I have 2 pi |
| 20:14.40 | lorsungcu_ | :/ |
| 20:14.57 | drmessano | I keep forgetting to order the SD cards |
| 20:15.03 | navaismo | lorsungcu, yes are correct too |
| 20:15.18 | navaismo | and not tested other script |
| 20:16.38 | Qwell | navaismo: The script is being launched fine. |
| 20:17.19 | Qwell | What makes you think your user has access to /sys/class/cpio/? |
| 20:17.48 | navaismo | if I use System cmd in dial plan it works |
| 20:17.59 | navaismo | if I run the script from linux shell it works too |
| 20:18.14 | navaismo | root is running asterisk |
| 20:20.36 | navaismo | but always I get this -- > "Failed to execute '/var/lib/asterisk/agi-bin/RasPiIO.agi': Permission denied" 1 |
| 20:22.18 | Qwell | pastebin the permissions of the script and of the agi-bin dir. |
| 20:23.50 | lorsungcu_ | also ps aux | grep asterisk |
| 20:23.52 | navaismo | all is 777 but 1 sec i pastebin |
| 20:27.12 | *** mode/#asterisk [-q igcewieling!*@*] by Qwell |
| 20:27.32 | *** mode/#asterisk [-q manxpower!*@*] by Qwell |
| 20:27.35 | navaismo | here http://pastebin.com/4iVtHkNM |
| 20:27.58 | Qwell | that doesn't properly show me the parent dir like I asked. |
| 20:28.05 | Qwell | (hint, it's not +r) |
| 20:30.38 | navaismo | Qwell, this? http://pastebin.com/UsaKDpN1 |
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| 20:38.44 | [TK]D-Fender | checkout time, BBIAB |
| 20:39.20 | navaismo | ruuning the scrip from cli only show a warning and works http://pastebin.com/0FQTuDKp |
| 20:39.52 | navaismo | the only thing for using phpagi is to get the state of the GPIO |
| 20:49.51 | mazpe | imma about to burn this audiocodes |
| 20:51.04 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 20:51.15 | cj | bah. I am having iax2 issues. Could someone help me figure this out? |
| 20:51.25 | lorsungcu_ | mb |
| 20:51.28 | cj | chan_iax2.c:10750 socket_process: Call rejected by <moo>: No authority found |
| 20:52.01 | cj | I have a hard time understanding registrar and regisgree entries in iax.conf |
| 20:52.03 | lorsungcu_ | psatebin conf files from both sides? |
| 20:53.38 | cj | http://paste2.org/p/2097722 |
| 20:53.40 | cj | that's on sip1 |
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| 20:55.02 | cj | http://paste2.org/p/2097723 |
| 20:55.05 | cj | that's on sip0 |
| 20:55.27 | lorsungcu_ | reverse usernames |
| 20:55.44 | cj | which? the register => ... or the [...] ? |
| 20:56.08 | lorsungcu_ | username= |
| 20:57.16 | cj | ok. can you explain to me how that works? I thought the [someuser] "context" set the username for the client registering with this host |
| 20:57.23 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
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| 21:00.57 | cj | well, I get a different error now. I guess that's better |
| 21:01.25 | lorsungcu_ | as i understand, you're basically setting the string you'd use to reach the destination by setting values under the [someuser] context |
| 21:01.47 | lorsungcu_ | so rather than dial(IAX/user:pass@whatever) |
| 21:01.58 | lorsungcu_ | the username/secret in the context is used |
| 21:02.05 | lorsungcu_ | what error do you see now/ |
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| 21:08.36 | cj | http://paste2.org/p/2097734 |
| 21:08.55 | cj | that's on sip0 |
| 21:13.19 | cj | man, I need a better way of getting those logs into pastebin |
| 21:13.23 | cj | checks out /var/log/asterisk |
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| 21:18.05 | lorsungcu_ | hmm |
| 21:21.09 | *** part/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 21:22.52 | lorsungcu_ | does the provider want 11 digits? |
| 21:22.58 | lorsungcu_ | er |
| 21:23.00 | lorsungcu_ | :p |
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| 21:30.24 | cj | I'd have to ask carrar. I'm going to stop hacking on this for the time being and take care of some other things. Thanks for your help ;-) |
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| 22:34.56 | ForEverAutistic | :) |
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| 22:35.36 | navaismo | still compiling argh |
| 22:36.11 | navaismo | only for a "denied" |
| 22:36.17 | navaismo | feels stupid |
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| 23:01.37 | *** join/#asterisk CrazyTux[m] (~Brandon@wsip-68-15-80-34.oc.oc.cox.net) |
| 23:01.49 | CrazyTux[m] | Is there an easy way in asterisk dialplan to check if a context exists (asterisk 1.4) |
| 23:02.20 | CrazyTux[m] | i.e. gotoif($[context_exists(context)) = "true"]?go here) |
| 23:11.30 | [TK]D-Fender | Nope |
| 23:18.08 | CrazyTux[m] | [TK]D-Fender: argh |
| 23:18.32 | CrazyTux[m] | [TK]D-Fender: doesn't this seem like critical routing / conditional functionality? |
| 23:21.11 | [TK]D-Fender | To see if a CONTEXT exists? No... why should that even be VARIABLE? |
| 23:21.20 | [TK]D-Fender | Its your dialplan.. youshould know if you even have a context.... |
| 23:21.45 | [TK]D-Fender | and I didn't sy there was NO way .... just nothing "easy |
| 23:22.30 | lorsungcu | could use a system() to check the file you expect the context to be in |
| 23:22.38 | lorsungcu | but tkd is right.. |
| 23:22.40 | CrazyTux[m] | [TK]D-Fender: It would help make it more dynamic. |
| 23:23.35 | CrazyTux[m] | It's also making me hack 100+ more lines to do some invalid extension context magic |
| 23:23.36 | CrazyTux[m] | lol |
| 23:23.41 | lorsungcu | how aer you planning to use that, tux? |
| 23:23.53 | [TK]D-Fender | If you're dealing with a text config and aren't worried about having actual "content" in there you could use BackTicks (1.4 add on) grepping your configs |
| 23:24.07 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 23:24.11 | CrazyTux[m] | I've several contexts setup like this: [star-code-69], [star-code-70], etc and they all have 's' exten |
| 23:24.39 | CrazyTux[m] | I'm just going to setup an i => and map each one... (crazy) |
| 23:24.40 | [TK]D-Fender | CrazyTux[m]: No need for 100's of lines. it is a tiny job to parse what got loaded. if ANYTHING exists in a context you can get that info |
| 23:24.49 | [TK]D-Fender | Yes... that IS crazy |
| 23:24.57 | [TK]D-Fender | but could work... |
| 23:25.09 | [TK]D-Fender | think about the times you DONT want those in there though... |
| 23:25.15 | CrazyTux[m] | [TK]D-Fender: well without a context_exists() this seems like the best solution ? |
| 23:25.32 | [TK]D-Fender | no, just parse out what did load. |
| 23:25.48 | CrazyTux[m] | [TK]D-Fender: so your saying to actually GREP the file? |
| 23:26.03 | CrazyTux[m] | That seems like it leaves lots of room for error |
| 23:27.22 | lorsungcu | still not clear on what you need to know about context existing for. |
| 23:28.38 | CrazyTux[m] | lorsungcu I want to know if I can route to context star-code-XX,s,1 if not "invalid extension action" |
| 23:28.39 | CrazyTux[m] | kind of deal |
| 23:28.56 | CrazyTux[m] | so I'm hacking around it, by mapping out EACH one right now.. :( |
| 23:29.06 | lorsungcu | any reason you dont just make them extensions? |
| 23:29.41 | CrazyTux[m] | lorsungcu I could do that too... but that means rewriting LOTS of code too |
| 23:29.45 | lorsungcu | eh |
| 23:29.54 | lorsungcu | it would be done properly :p |
| 23:29.55 | CrazyTux[m] | lorsungcu so im hacking it, and making extensions that map to contexts |
| 23:29.59 | CrazyTux[m] | ;) |
| 23:30.34 | lorsungcu | i guess i dont see the advantage there. |
| 23:30.38 | [TK]D-Fender | CrazyTux[m]lorsungcu I want to know if I can route to context star-code-XX,s,1 if not "invalid extension action" <- that is a specific EXTENSION, not a "context" |
| 23:30.43 | [TK]D-Fender | that is not at all what you first asked |
| 23:31.32 | [TK]D-Fender | And is something you CAN do easy in the dialplan itself |
| 23:31.39 | CrazyTux[m] | [TK]D-Fender: listening :) |
| 23:31.53 | CrazyTux[m] | has already typed this out but listening :P |
| 23:32.03 | lorsungcu | its exactly what the i=> extension is there for... |
| 23:32.15 | [TK]D-Fender | show application chanisavail <- |
| 23:32.38 | CrazyTux[m] | lorsungcu when you Goto() an invalid context it goes into limbo and call hangs up, it does not fall into invalid exten processing |
| 23:32.58 | lorsungcu | right, but what you have made contexts should be extensions. |
| 23:33.20 | CrazyTux[m] | Unless of course I could do a dynamic context ;) then this would also be solved easier. |
| 23:33.27 | CrazyTux[m] | [TK]D-Fender: is this from the CLI ? |
| 23:33.36 | lorsungcu | yes |
| 23:33.39 | CrazyTux[m] | [TK]D-Fender: I would want to do it in dialplan |
| 23:33.51 | [TK]D-Fender | .... |
| 23:33.56 | CrazyTux[m] | ahh nm |
| 23:33.59 | CrazyTux[m] | apologies :) |
| 23:34.02 | CrazyTux[m] | [TK]D-Fender: I see :) |
| 23:34.04 | [TK]D-Fender | CLI command to get DIALPLAN APP INSTRUCTIONS. |
| 23:47.01 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 23:47.01 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 23:47.48 | CrazyTux[m] | [TK]D-Fender: it returns a status of zero no matter if it exists or not? it also returns the name regardless? |
