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| 04:29.04 | lxsameer | hi can I use AGIs to change the flow based on some condition and play some sounds or call a foreign number via AGI ? |
| 04:31.40 | ChannelZ | You can do just about anything you can in a dialplan, but via whatever language you like via AGI |
| 04:31.47 | ChannelZ | So yes |
| 04:32.58 | lxsameer | ChannelZ: cool, so can I use Ruby too |
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| 04:46.42 | ChannelZ | anything that you can do stdin and stdout with sure |
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| 09:04.37 | juanmapalad | ping |
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| 09:45.23 | ipalmer | morning all, I have 2 different databases on my server which I need to do joins across whilst using func_odbc. I can use different dsn's whilst using one particular database but not when I'm doing cross database joins, is this possible in Asterisk? |
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| 09:50.35 | skrusty | ipalmer: not that i know of |
| 09:51.40 | skrusty | what db server? mysql/mssql? |
| 09:51.45 | ipalmer | mysql |
| 09:52.14 | skrusty | :/ i know mssql supports linked servers, i don't think mysql does |
| 09:52.32 | ipalmer | i may have to federate the table onto the single database and do it that way |
| 09:53.06 | skrusty | or you could use curl and get the result from some service that cna talk to both dbs |
| 09:53.10 | skrusty | like a rest api |
| 09:53.59 | ipalmer | ok, not used curl before I'll take a look at that |
| 09:54.19 | skrusty | just throwing out ideas :) |
| 09:54.39 | ipalmer | cheers, appreciate all the ideas i can get |
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| 09:57.29 | earthworm | Hi there, I installed AsteriskNOW, and set it all up, I then decided to do a "yum update" and now the FreePBX UI is having problems. |
| 09:57.55 | earthworm | I then Googled and found there are official upgrade scripts. I rolled back the YUM transaction and then ran the proper scripts. |
| 09:58.26 | earthworm | Can anyone help with this? |
| 09:58.58 | earthworm | I'm not sure whether it's better to ask here or #freepbx |
| 10:01.51 | skrusty | maybe, sounds quite specific |
| 10:02.07 | skrusty | ask in there, no harm! :) |
| 10:02.11 | wdoekes | ~freepbx |
| 10:02.11 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
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| 10:26.26 | asteriskbizua | Greeting |
| 10:27.03 | asteriskbizua | where can i find desc about available channel states |
| 10:27.17 | asteriskbizua | ChannelStateDesc Down Rsrvd OffHook Dialing Ring Ringing Up Busy Dialing Offhook Pre-ring Unknown |
| 10:27.56 | asteriskbizua | i need desc about each of listed state |
| 10:28.04 | asteriskbizua | thanks a lot |
| 10:37.00 | skrusty | google? http://bit.ly/1qftXu9 |
| 10:37.07 | skrusty | first link, about half way down :) |
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| 11:51.19 | BakaKuna | ping |
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| 13:12.01 | bananapie | Is there a command I can run to disable all transcoding on asterisk so I can test my sound files to make sure all my menus are in the right format? |
| 13:13.17 | [TK]D-Fender | no |
| 13:13.36 | [TK]D-Fender | And what is "the right format"? |
| 13:15.12 | bananapie | the right format means that I have the files in g711, and g722 . depending on the call, the right format is used, and asterisk doesn't need to any real work. |
| 13:15.23 | bananapie | to do* |
| 13:16.33 | file | there is no command but if you don't load the modules which do the transcoding for the codecs then it won't |
| 13:17.06 | bananapie | ok. So I start up asterisk with no codecs and try and call my menu. I'll have to do this outside of business hours. thanks file |
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| 13:18.38 | [TK]D-Fender | bananapie: Or you could just lOOK at the files and see that you have one in every format and STAT it. |
| 13:20.07 | bananapie | I already checked the files are in each format, but I like being thorough. |
| 13:22.01 | *** join/#asterisk jkroon (~jkroon@dustpuppy.is.co.za) |
| 13:22.19 | jkroon | hi all, is there a way to execute a dialplan application/function directly from the CLI? |
| 13:27.56 | Katty | eyes bananapie with a spoon. |
| 13:28.17 | bananapie | What?!?!?! |
| 13:28.20 | bananapie | is evolving.... |
| 13:28.26 | Katty | :< :< |
| 13:28.42 | Katty | gah!!! |
| 13:28.52 | moldyshoe | used confusion |
| 13:29.00 | Katty | looses 1 hp |
| 13:29.04 | Katty | goes back to normal |
| 13:29.19 | moldyshoe | ;) |
| 13:29.33 | Katty | moldyshoe gains 1 xp! |
| 13:29.37 | Katty | moldshoe is now level 2! |
| 13:29.40 | Katty | hp goes up by 2! |
| 13:29.42 | moldyshoe | is devolving... |
| 13:29.59 | Katty | now i wanna play earthbound. |
| 13:30.31 | [TK]D-Fender | jkroon: Nope |
| 13:30.53 | [TK]D-Fender | jkroon: You COULD create some dialplan and originate a channel to use it however |
| 13:30.55 | bananapie | jkroon, you can request a hangup of a channel from the console, |
| 13:32.34 | jkroon | [TK]D-Fender, yea, that's what I normally do. In this case I need to escape from the asterisk CLI to a shell... |
| 13:32.55 | [TK]D-Fender | jkroon: huh? |
| 13:34.02 | jkroon | [TK]D-Fender, don't ask. busy with stuff that I normally don't want to get involved with. Off to the DC then to go and hack it at the terminal. Has a client where a previous provider set stuff up for him and now refuses to hand over the keys. |
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| 14:18.21 | kb3ien | anyone had success auto-provisioning snom320 with DHCP ? |
| 14:19.01 | WIMPy | Yes. |
| 14:19.13 | WIMPy | Hope that answer helps. |
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| 14:21.22 | kb3ien | Tried just option tftp-server-name with a URI after it, but its not being read. |
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| 14:24.05 | WIMPy | Does it request the file? |
| 14:24.13 | WIMPy | is using http. |
| 14:25.25 | kb3ien | It never even contacts port 80, yes. I've tried tftp u68 and http t80, neither port is contacted. |
| 14:26.18 | kb3ien | I tried to do this: http://pastebin.com/eEjrvxaC only simpler. |
| 14:28.08 | kb3ien | it does reach out to provisioning.snom.com via http though. |
| 14:28.46 | WIMPy | Does it use dhcp at all? Have you tried to do a factory reset? |
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| 14:33.43 | kb3ien | i see it gets it's ip address from dhcpd |
| 14:35.16 | kb3ien | trying that now... |
| 14:36.21 | kb3ien | i feel stupid. Apperently a factor reset is the kind of reset they DIDNT give it at the factory... |
| 14:37.06 | kb3ien | s/factor/factory/ |
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| 15:05.17 | pjensen00 | I'm using ARI and I've been able to connect to the web socket, create a bridge, destroy a bridge, answer a channel but I fail to add a channel to the bridge. I get a 204 (success) back from the command but the logs indicate the channel fails to join. Unfortunately there is no indication of WHY the channel failed to get added to the bridge. Anyone have any good insight based off this log dump? http://pastebin.com/0xUZUGS1 |
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| 15:34.53 | mjordan | pjensen00: what is role=Default? |
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| 15:36.07 | mjordan | hm. We need to get the allowed roles documented better in the auto-generated wiki docs |
| 15:36.09 | *** part/#asterisk sklav (~sklav@unaffiliated/sklav) |
| 15:36.11 | mjordan | So, "Default" is not a valid role |
| 15:36.20 | *** join/#asterisk gnudna (~sklav@unaffiliated/sklav) |
| 15:36.23 | mjordan | just remove the role parameter that you're passing. You most likely don't need it |
| 15:36.50 | mjordan | It's a fairly specialized field that is really only useful in holding bridges when you want a channel to act as an announcer to the other holding bridge participants |
| 15:37.11 | mjordan | At least, today. In the future, other bridging types could have other more complex role types |
| 15:38.07 | *** join/#asterisk navaismo (0c0adbe1@gateway/web/freenode/ip.12.10.219.225) |
| 15:38.22 | gnudna | <PROTECTED> |
| 15:38.46 | WIMPy | That's what usually happens. |
| 15:39.03 | WIMPy | Unless you explicitely told it to answer first. |
| 15:39.30 | gnudna | i guess my config must have an issue somewhere it seems to answer and then to route |
| 15:39.40 | navaismo | using a GUI? |
| 15:39.43 | gnudna | can i paste my extensions.conf for you to take a look? |
| 15:39.46 | gnudna | no |
| 15:39.53 | gnudna | debian asterisk from backports |
| 15:39.59 | gnudna | so im runnin 11.x |
| 15:40.18 | WIMPy | Either you have an explicit Answer() or you Playback() something. |
| 15:40.41 | gnudna | i do have an answer() in default |
| 15:40.51 | gnudna | but for my inboud rule i do not |
| 15:40.53 | WIMPy | Guess what that does... |
| 15:41.00 | gnudna | ;) |
| 15:41.05 | WIMPy | ok |
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| 15:41.51 | gnudna | that being said if i post my config care to offer me some good advice? |
| 15:41.51 | WIMPy | Do the calls actually arrive in the context you expect? |
| 15:42.05 | gnudna | I thought so |
| 15:42.08 | WIMPy | Just do it. |
| 15:42.20 | WIMPy | You will find out who has comments about it then. |
| 15:42.21 | navaismo | i saw many threads recently talking about that(the answer) maybe there is a guide there telling that... idk |
| 15:42.53 | gnudna | did follow a guide when i initially set this up |
| 15:43.07 | [TK]D-Fender | gnudna: it's your dialplan.. do whatever you want |
| 15:46.48 | lvlinux | If I'm using mg2 echo canceller and have "echotraining=yes" why would I be getting the message "chan_dahdi.c:5067 dahdi_train_ec: Unable to request echo training on channel 3: Invalid argument" |
| 15:47.04 | lvlinux | on the cosole when a call is placed. |
| 15:47.09 | lvlinux | ? |
| 15:47.57 | gnudna | WIMPy http://pastebin.com/BfAiLAb2 |
| 15:48.19 | [TK]D-Fender | gnudna: show us the call |
| 15:49.00 | gnudna | what call [TK]D-Fender ? |
| 15:49.28 | [TK]D-Fender | gnudna: the call where it isn't acting the way you want it to |
| 15:49.49 | [TK]D-Fender | [11:37]gnudnaHi guys is there a way for asterisk to not answer a call before it performs its for lack of a better word routing <- show us THIS call |
| 15:50.59 | gnudna | ok give me a minute to get a new call going. but im pretty sure my extensions.conf has an issue that i pasted above |
| 15:51.28 | [TK]D-Fender | gnudna: Several issues |
| 15:51.36 | gnudna | ;) |
| 15:51.45 | gnudna | care to clarify so i can fix them please |
| 15:52.57 | [TK]D-Fender | patterns that clearly don't match what you think you're going to dial. misformatted dial's for multiple devices, cramming inbound & outbound extens in the SAME context. |
| 15:53.53 | gnudna | ok i thought they were each isolated |
| 15:55.04 | [TK]D-Fender | [users] <-- you have inboutn and outbound stuff in there |
| 15:55.07 | [TK]D-Fender | BIG mes |
| 15:55.09 | [TK]D-Fender | ess* |
| 15:55.11 | [TK]D-Fender | mess* |
| 15:55.29 | gnudna | ok |
| 15:55.58 | gnudna | like i said followed a popular howto when i set this up |
| 15:56.41 | [TK]D-Fender | gnudna: which? |
| 15:56.43 | gnudna | so where should i include those into? [default] |
| 15:56.57 | [TK]D-Fender | you should never have a context even named [default]. EVER |
| 15:57.36 | [TK]D-Fender | you will have contexts for your inbounds, and context for your devices taht are supposed to dial out. Perhaps a heirarchy of them |
| 15:57.47 | gnudna | fair enough that is based on default asterisk config that shipped |
| 15:58.06 | [TK]D-Fender | that's the entire point of these; so taht calls taht come in can just get redirected out due to a small mistake |
| 15:58.15 | [TK]D-Fender | gnudna: that isn't a default config. |
| 15:58.34 | [TK]D-Fender | gnudna: that is a sample SYNTAX REFERNCE. that is not necessarily a sane way to configure a system |
| 15:58.50 | gnudna | fair enough |
| 15:59.31 | gnudna | now based on what you said and what i posted how can i re-organize it work properly and securely |
| 15:59.46 | [TK]D-Fender | spit it up |
| 15:59.49 | [TK]D-Fender | split* |
| 16:00.20 | lvlinux | lol |
| 16:01.10 | gnudna | ok do the includes need to be defined? |
| 16:01.35 | [TK]D-Fender | do you need what's in them to be included? |
| 16:01.59 | gnudna | i would think yes to make outbiund calls |
| 16:02.19 | [TK]D-Fender | Go split up your inbound & outbound |
| 16:02.45 | gnudna | they are already split out |
| 16:02.52 | gnudna | seems im missing something here |
| 16:04.05 | [TK]D-Fender | you have a context named [default] with ALL of that crap in there |
| 16:05.05 | gnudna | all i see is [default] with one line below it before the next context is defines |
| 16:05.14 | gnudna | unless it's a top down inclusion |
| 16:07.20 | [TK]D-Fender | gnudna: Sorery, bad aim .. [users] |
| 16:07.36 | [TK]D-Fender | gnudna: [default] should just be removed completely |
| 16:11.04 | gnudna | ok |
| 16:11.16 | gnudna | done |
| 16:11.57 | gnudna | now what else needs cleaning or removal? |
| 16:12.20 | [TK]D-Fender | gnudna: Go try some calls and see what lands where. |
| 16:12.27 | gnudna | ok |
| 16:12.49 | [TK]D-Fender | gnudna: http://pastebin.com/BfAiLAb2 <- what are lines 23-27 doing there? |
| 16:13.22 | gnudna | i thought they registering the extensions listed |
| 16:13.34 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:13.34 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:13.38 | gnudna | our routing the extensions listed |
| 16:13.55 | [TK]D-Fender | gnudna: .... look at those lines |
| 16:14.22 | [TK]D-Fender | gnudna: exten = s. What do you have pointed to that contedxt that will lead to "s" ever getting used? |
| 16:14.57 | gnudna | ;) noting that i am aware off |
| 16:15.01 | gnudna | nothing |
| 16:15.20 | gnudna | ok so i can clean that out |
| 16:16.27 | gnudna | i removed them made their way in when i was testing something from work |
| 16:16.31 | [TK]D-Fender | gnudna: You need to actually look at what you're doing. |
| 16:20.58 | gnudna | [TK]D-Fender so far all is working as expected and i removed what you pointed out because it is not being used as you pointed out |
| 16:21.16 | gnudna | im not just copying and pasting |
| 16:21.26 | Penguin | s does not mean "everything else" |
| 16:21.59 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:22.10 | Penguin | I remember conceiving that, based on things I read on the internet, when I started using asterisk. |
| 16:22.50 | gnudna | any other glaring errors [TK]D-Fender |
| 16:23.18 | gnudna | yeah i followed a bunch of how-to etc.. and assumed i understood what it was doing |
| 16:23.28 | pjensen00 | mjordan: Awesome! That worked |
| 16:24.10 | Penguin | Since that time, many others have shown up here with the same misconception that s matches anything not explicitly defined. |
| 16:25.08 | WIMPy | When I started I read that s is the start extension, thinking that would be something that is run before anything else. A feature I'm missing since I found out it isn't. |
| 16:25.27 | gnudna | ;) |
| 16:25.59 | Penguin | The good thing is that you could configure it to use s first. There's no way to make s match everything that doesn't otherwise match. |
| 16:26.05 | Penguin | So you've got an advantage. |
| 16:26.22 | WIMPy | How could I do so? |
| 16:26.38 | Penguin | Start by directing all your calls to extension s. |
| 16:27.11 | gnudna | ok and im i at least setting the call display correctly? |
| 16:27.15 | WIMPy | By always dialling s first? |
| 16:27.41 | Penguin | Right. If you send calls to s, calls will start at s. Exten s will then send your calls to other extens. |
| 16:27.52 | Penguin | It's not practical, but it is possible. |
| 16:28.13 | WIMPy | Indeed. Not practical at all. |
| 16:28.28 | Penguin | But making s a catch-all exten... not going to work. |
| 16:28.54 | WIMPy | Easy. |
| 16:29.00 | pjensen00 | mjordan: Also, is there a reason to not use the holding bridge type? From what I read on the Bridges page the bridge type in A12 is automagic from a user's perspective now |
| 16:29.13 | WIMPy | exten i,1 -> Goto(s,1) |
| 16:29.35 | Penguin | But exten i is not called directly, and it is not called when no other exten matches. |
| 16:29.52 | Penguin | i is only called from other applications. |
| 16:30.34 | Penguin | For instance, if you dial the phone number when trying to make a call, it does not hit i. |
| 16:30.52 | Penguin | s/the/the wrong/ |
| 16:30.54 | Penguin | dammit |
| 16:30.56 | WIMPy | Depends on the channeltype. |
| 16:31.05 | Penguin | Assume SIP. |
| 16:31.22 | WIMPy | Why would I? |
| 16:31.27 | Penguin | That's the "normal" tech for all VoIP, and especially asterisk. |
| 16:31.39 | WIMPy | Always expect the worst? |
| 16:31.58 | Penguin | That would also be a good idea. :) |
| 16:32.34 | WIMPy | Since using Asterisk, I know that my imagination is too limited to do so. |
| 16:33.22 | WIMPy | But Linux has been quite evil lately, as well. |
| 16:33.47 | gnudna | [TK]D-Fender aside from what has been pointed out anything else obviously wrong? |
| 16:34.58 | [TK]D-Fender | Go look at everything you made and ask yourself: who can dial it? When? Is the pattern sane and specifically cover what I want? Am I covering ALL of what I should be doing? |
| 16:35.32 | WIMPy | Noty to forget: Did I accidentally cover more than I wanted? |
| 16:36.15 | [TK]D-Fender | noted a few "mores" in there... |
| 16:36.23 | gnudna | [TK]D-Fender i believe i am but obviously i was wrong so im asking for a second opinion |
| 16:36.58 | *** join/#asterisk navaismo (0c0adbe0@gateway/web/freenode/ip.12.10.219.224) |
| 16:37.29 | WIMPy | The most obvious one is that you most probably don't want to allow outbound calls in your default context. |
| 16:37.34 | [TK]D-Fender | gnudna: what is [phoneservices] for? |
| 16:37.51 | gnudna | to catch 311 411 etc... |
| 16:38.03 | [TK]D-Fender | gnudna: what does that pattern match? |
| 16:38.13 | [TK]D-Fender | gnudna: is 555 legal? |
| 16:38.18 | [TK]D-Fender | gnudna: it is there.... |
| 16:38.34 | [TK]D-Fender | gnuthat allows MORE than you want. |
| 16:39.08 | gnudna | i know that i did not want to explicitily define 911 etc... |
| 16:39.20 | gnudna | i could if there was a security reason do to so |
| 16:39.37 | [TK]D-Fender | gnudna: 411 costs money most of the time... |
| 16:39.43 | [TK]D-Fender | gnudna: SURE you want to allw that one? |
| 16:40.00 | gnudna | i know but it is blocked on the voipms portal |
| 16:40.08 | gnudna | so im not worried |
| 16:40.09 | [TK]D-Fender | gnudna: what if there is a BAD possible value? Go find out EXACTLY what you want it to cover and make something to match those |
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| 16:40.28 | [TK]D-Fender | gnudna: you are playing a game called "Lucky". It runs out after a while |
| 16:41.28 | gnudna | so i best define all the rules explicitly which i will do so shortly |
| 16:41.56 | gnudna | i did not think it was a concern really since this is a home phone |
| 16:43.02 | [TK]D-Fender | gnudna: Not thinking is not a good approach. ebfore you know it you let random outside callers get through and start placing calls on your system where you would never want to even dial for yourself |
| 16:43.46 | gnudna | fair enough and i agree the practice is wrong |
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| 16:44.17 | gnudna | but not thinking is kinda harsh since up till this moment i thought it was the valid way |
| 16:44.58 | WIMPy | If you think you think, you only think you think. |
| 16:45.21 | gnudna | ego is bruised |
| 16:46.59 | gnudna | now you mentioned outbound in the default context WIMPy ? |
| 16:47.02 | navaismo | let the phone´s bill raise to 10000USD! because #YOLO |
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| 16:47.51 | gnudna | navaismo can i have your CC then? |
| 16:48.17 | gnudna | ;) |
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| 16:53.20 | gnudna | that said guys any other obvious issues before i call it a day aka lunch break |
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| 16:53.57 | kb3ien | is the valet app still the best way to park and annouce with BLFs for the parking lot? |
| 16:54.10 | kb3ien | or are my notes OLD? |
| 16:54.50 | gnudna | [TK]D-Fender thank you for all your help. |
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| 17:05.03 | wasanzy | hello |
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| 17:36.02 | navaismo | o/ |
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| 17:37.10 | gmalsack | hey all!!! happy thursday. |
| 17:37.35 | gmalsack | running ast 12. anyone know of a way to set the default allowed codecs globally in pjsip? |
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| 17:51.53 | file | gmalsack, are you using a config file? |
| 17:52.05 | gmalsack | file: yup |
| 17:52.12 | file | then use config templates |
| 17:52.35 | file | https://wiki.asterisk.org/wiki/display/AST/Using+Templates?src=search |
| 17:52.44 | gmalsack | awesome! thanks.... |
| 17:55.53 | gmalsack | sweet!!! works like a charm! |
| 17:56.37 | gmalsack | so did they do away with [default] or the like in pjsip.conf? |
| 17:57.21 | file | yes. |
| 18:00.47 | gmalsack | file: is pjsip the only major overhaul in 12 vs 11? |
| 18:01.00 | file | uh, no |
| 18:01.10 | pjensen00 | lots different |
| 18:01.28 | file | I suggest you read the CHANGES and UPGRADE file |
| 18:01.47 | gmalsack | ugh.... guess I'll have to start looking at the docs and get up to speed on it.... :-( like I had nothing better to do!!! lol |
| 18:01.56 | gmalsack | Downloading them now. |
| 18:02.39 | drmessano | I'm going to stay on Asterisk 11 forever |
| 18:03.42 | gmalsack | drmessano: lmao!!!! right.... |
| 18:04.00 | Qwell | drmessano: You upgraded from CVS 0.9? nub |
| 18:05.17 | gmalsack | ugh, I've been working on a new build of a system for over a year now for our company. it's comprised of 6 servers, and is designed to simply be able to "plug in" an additional asterisk box as needed, and automatically remove failed boxes.... biggest pain in my ass is replacing the flash operator panel that the company uses religiously.... |
| 18:05.58 | WIMPy | FOP for a system that needs six servers? |
| 18:06.12 | gmalsack | yea.... fun hey! |
| 18:06.25 | WIMPy | That sounds ... impressive. |
| 18:06.26 | gmalsack | they are not all asterisk boxes. |
| 18:07.16 | gmalsack | 1, opensips, 2 asterisk, 1 mysql, 1 storage, 1 web-app server. then as the load builds, just add more and more asterisk boxes.... :-) |
| 18:11.39 | [TK]D-Fender | [14:03]Qwelldrmessano: You upgraded from CVS 0.9? nub <- Those were the days.... </bunker> |
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| 18:37.00 | circ-user-AHpid | it is possible to use dynamic features in queues ? ... i am trying, but it is not working ... can anyone guide me? |
| 18:39.59 | [TK]D-Fender | Where in queues? |
| 18:41.37 | josemslopes | when the caller is waiting |
| 18:42.01 | [TK]D-Fender | No |
| 18:43.33 | lvlinux | are software echo canceller names in system.conf case sensitive? i.e. does it matter if I use mg2 or MG2??? |
| 18:43.49 | Qwell | yes |
| 18:45.03 | lvlinux | which is correct then? |
| 18:46.15 | Qwell | mg2 |
| 18:46.37 | lvlinux | oh ok. well that's not my problem then... trying to get this horrible echo under control... |
| 18:47.10 | *** part/#asterisk Eugene (eugene@kashpureff.org) |
| 18:47.21 | lvlinux | people using the phones say it's about the same no matter what I do... |
| 18:47.40 | josemslopes | about dynamic features on queue. what i need is, when a caller is waiting on the queue, the caller can press a key to ear the hold time (instead of the announce of holdtime ). Can anyone give me a sugestion? |
| 18:48.54 | WIMPy | hold time? |
| 18:50.00 | josemslopes | yes |
| 18:50.16 | [TK]D-Fender | josemslopes: go patch app_queue.c because * doesn't have an option for that |
| 18:50.42 | WIMPy | They most probably have that on their display. |
| 18:54.59 | josemslopes | the hold time that i mean is the estimated wait time to the caller be answered |
| 18:55.27 | mjordan | josemslopes: you could build something that does that using various Asterisk APIs, but nothing in Asterisk is going to provide that to you natively. |
| 18:55.45 | navaismo | lvlinux: use another echo canceller or use hardware, you have OSLEC, HPEC from Digium OCtasic from openvox |
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| 18:56.16 | [TK]D-Fender | mjordan: Which would allow another process to both spy on that channel, and detect dtmf from it, etc> |
| 18:57.20 | lvlinux | navaismo: well i tried oslec and they say it's no different. I only went back to mg2 because it allows up to 1024 taps---and there is still echo. "dahdi show channel x" shows that the cancellation is enabled during a call. This is on a Sangoma card, with no hardware EC :-( |
| 18:58.10 | lvlinux | ^^^ "they" meaning the people using the phones (long way away from me---I'm doing this remotely) |
| 18:58.27 | josemslopes | mjordao: what i need is to asterisk capture the dtmf in queue (or spy like [TK]D-Fender said) and let me to run a command or a dynamic feature ... |
| 18:58.32 | mjordan | [TK]D-Fender: yup. AMI + some Local channels and maybe a ChanSpy would probably do the trick. |
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| 18:58.49 | mjordan | Maybe. It'd require some hackery. Or you could just use ARI and build the app yourself :-D |
| 18:58.53 | [TK]D-Fender | mjordan: Didn't think it could bridge DTMF... |
| 18:58.59 | mjordan | It can listen for DTMF |
| 18:59.09 | [TK]D-Fender | Then that might do. |
| 18:59.27 | WIMPy | Local channels? Wouldn't a chanspy be enough? |
| 18:59.29 | mjordan | The trick is playing the announcement and then getting them back into the Queue (if you remove them from Queue to play the announcement) |
| 18:59.34 | mjordan | WIMPy: Probably |
| 18:59.48 | [TK]D-Fender | But that is a NSATY hack, and you'd need to prevent it from running once the user is connected, or leaves the queue, etc |
| 18:59.54 | [TK]D-Fender | This would be a NASTY job. |
| 19:00.47 | WIMPy | Actually it doesn't sound that bad. The issue I see is that you won't easily be able to stop MOH while plaing the time. |
| 19:01.23 | [TK]D-Fender | yup |
| 19:01.56 | [TK]D-Fender | I don't see the value in letting some neurotic caller press a button to hear a lie repeatedly. |
| 19:01.59 | WIMPy | Unless you use silence in the queue and play MOH via the chanspy. |
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| 19:02.47 | mjordan | By the way, this is the reason why we've been spending time on ARI :-) |
| 19:03.03 | mjordan | Once you get into "I wish Asterisk did this crazy thing", you're better off writing your own Queue app and making it do whatever you want |
| 19:03.25 | WIMPy | And why do I need ARI for that? |
| 19:03.30 | mjordan | It's just easier. |
| 19:03.44 | WIMPy | Easier than AMI? |
| 19:03.47 | mjordan | I think so |
| 19:04.18 | WIMPy | Having to sent statless requests doesn't sound easier to me. |
| 19:04.35 | WIMPy | Or maybe I didn't get the concept (yet). |
| 19:04.50 | mjordan | Think about it this way: with both AMI/ARI, things are asynchronous. You get events telling you what is going on. |
| 19:05.02 | mjordan | Unlike AMI, ARI let's you manipulate the 'building blocks' that applications are built on top of. |
| 19:05.13 | mjordan | Say I wanted to move a channel from a multi-party bridge to a waiting area that plays MoH |
| 19:05.28 | mjordan | In AMI, I'd have to have an extension that has a ConfBridge. I'd redirect them to an extension that Parks. |
| 19:05.40 | mjordan | I'd have to configure my Park and ConfBridge so that they didn't appear like a ConfBridge or Parking |
| 19:05.46 | mjordan | And hopefully get the behaviour that I want |
| 19:05.55 | mjordan | I'd have to configure my dialplan so that it doesn't do weird things to the channel while I move it around |
| 19:06.05 | mjordan | And I'd be doing Redirects, which are - often times - evil (tm) |
| 19:06.23 | mjordan | With ARI, I just do a removeChannel on my mixing bridge, and an addChannel on my holding bridge. Done. |
| 19:06.28 | WIMPy | So the only ting that was missing was a way to launch an application via AMI instead of having to use the dialplan? |
| 19:06.57 | mjordan | With ARI, you are writing the application that is called from the dialplan. You're manipulating all the things that we typically manipulate in C. |
| 19:07.12 | mjordan | With AMI, you're moving things around in the dialplan. |
| 19:08.01 | mjordan | Often, with AMI + AGI, you can do 95% of the things you can do with ARI - that's true. It's just harder. |
| 19:08.13 | WIMPy | Ok, so you have more control, but AMI still appears much easier to me. |
| 19:08.18 | mjordan | that's fair |
| 19:08.25 | mjordan | I find RESTful things to be easier, personally :-) |
| 19:08.58 | WIMPy | I like reliable connections. |
| 19:09.00 | *** join/#asterisk mirela666 (~mirko.bra@95.180.116.173) |
| 19:09.15 | mjordan | WebSockets are reliable. |
| 19:09.29 | mjordan | Asterisk's HTTP server supports persistent connections (now), which is based of course on TCP. |
| 19:10.15 | WIMPy | Ok, so if it's persistent, I guess the difference isn;t that big any more. |
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| 21:01.11 | pjensen00 | .... question. Will using an active Stasis application connected to a websocket block ALL other AGI applications from executing? |
| 21:02.18 | pjensen00 | I have multiple waiting AGI based programs that get handed a channel, but none of them get anything if I hand a single one of them over to an active Stasis application |
| 21:07.32 | smkelly | file: hi |
| 21:07.41 | file | smkelly, hi |
| 21:07.48 | smkelly | file: hi |
| 21:32.58 | pjensen00 | HA! Nevermind, I am silly. I have a load balancer issue absorbing the second call |
| 21:33.04 | pjensen00 | I knew the problem didn't make sense at face value |
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| 22:38.49 | mjordan | pjensen00: Yup :-) AGI and ARI are on two completely separate "paths", so to speak |
| 22:40.13 | pjensen00 | mjordan: I figuted it was something obvious because it made no sense why one would affect the other. |
| 22:40.49 | pjensen00 | Though for some reason, "Bridge show all" is stuck and it doesn't update when bridges are being created/destroyed. |
| 22:41.05 | pjensen00 | I think asterisk is wagging its finger at me for trying to be creative. |
| 22:45.39 | mjordan | pjensen00: that is odd. The CLI command shouldn't get 'stuck'. |
| 22:46.02 | mjordan | pjensen00: may be worth snagging a backtrace, particularly of the gdb variety |
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| 23:24.11 | masterbase | Hi, everybody! I trying to confirm a security issue I discoverd. If you initialize IAX2 keys via "keys init" the passphrase gets stored in plaintext to the .asterisk_history file in the users home directory. Can anybody confirm this? |
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