| 00:10.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 00:26.31 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.0.1 (2014/11/20), 11.14.1 (2014/11/20), 1.8.32.1 (2014/11/20); Standard: 12.7.1 (2014/11/20); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 00:26.45 | *** join/#asterisk antiochIst (~taylorhaw@168.244.48.230) |
| 00:35.25 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 00:36.52 | *** join/#asterisk Milenco (~Milenco@home.milenco.net) |
| 00:38.52 | krapper | anyone provide some guidance on the following... SIP/blah-1234 connected line has changed. Saving it until answer for SIP/trunk-01234 |
| 00:41.17 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 00:42.36 | Milenco | what do you want to know? |
| 00:42.53 | Milenco | its usually the stay its going into before answering the call |
| 00:43.34 | Milenco | usually its the ringing state |
| 00:43.39 | krapper | Milenco, JW was just looking for further clarification on what's happening and if something that should be addressed? |
| 00:44.26 | Milenco | no, its just a debug message |
| 00:46.28 | krapper | ok.. thank you |
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| 01:18.01 | WIMPy | Milenco: So you're from the late shift? |
| 01:18.32 | Milenco | hm? |
| 01:18.49 | Milenco | no im at home, just exploring asterisk some more |
| 01:18.56 | WIMPy | Just looking at the clock. |
| 01:19.16 | Milenco | need to setup a multi-queue/ivr callcenter systems in a few months |
| 01:19.26 | Milenco | already doing asterisk maintenance for a year or so |
| 01:19.27 | WIMPy | So more like the private night shift then. |
| 01:19.44 | Milenco | but designing and setting up new stuff is pretty new to me |
| 01:19.55 | Milenco | also reading the definitive guide now |
| 01:20.17 | WIMPy | If you hang out here, there's also #asterisk-de, btw. |
| 01:20.35 | Milenco | tba ms |
| 01:20.39 | Milenco | ehh..i mean thanks |
| 01:20.42 | Milenco | but im Dutch :) |
| 01:21.05 | WIMPy | Ok, almost :-) |
| 01:21.09 | Milenco | Our company aquired a German competitor and now we want to deploy our phone system there as well |
| 01:21.17 | Milenco | haha some might consider that an insult! |
| 01:21.29 | Milenco | but i live 10km near the border anyway :P |
| 01:21.42 | Milenco | what about you WIMPy? |
| 01:22.03 | WIMPy | If we ever meet, I'll tell you my favourite joke about a german and a dutch guy :-) |
| 01:22.24 | WIMPy | Right next to Denmark. |
| 01:22.28 | Milenco | somehow im already looking forward to it :) |
| 01:23.04 | Milenco | what is your professional relation with asterisk? |
| 01:25.15 | WIMPy | Well, I first heard about it 2002, but decided it wasn't usable. The at some time when 1.4 was jsut released I started to play with it for doing some value added services. |
| 01:25.46 | WIMPy | Now I do small IVR/VM addons to the PBXs of small companies. |
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| 01:26.07 | Milenco | nice :) you might say you are an asterisk specialist :) |
| 01:26.22 | WIMPy | In some areas. |
| 01:26.52 | Milenco | i inherited an multi-location asterisk environment with about 500 users |
| 01:26.56 | WIMPy | Like constantly moaning about PSTN/ISDN integration. |
| 01:27.06 | Milenco | from 7 locations all over the world, most europe |
| 01:27.17 | WIMPy | Nice |
| 01:27.31 | Milenco | I've been tasked with several new Asterisk tasks as our previous 'main' Asterisk guy left |
| 01:27.56 | Milenco | There is lot of room for improvement and I want to do it 'nice' so reading up a lot |
| 01:28.39 | WIMPy | If you want to make it nice, it easily becomes a full time job, I think. |
| 01:29.02 | Milenco | true, but i constantly see how i can do things better |
| 01:29.19 | Milenco | last month i rewrote our callout macro to include several voip/isdn failovers |
| 01:29.24 | WIMPy | As long as that works, it's the good days :-) |
| 01:29.37 | Milenco | but i look at it now and think: i can do that with half the code and a much better layout |
| 01:30.32 | WIMPy | Do it. Less shit makes it easiert to keep track of things. |
| 01:31.02 | Milenco | yeah, gonna try to rewrite most things anyway |
| 01:31.07 | Milenco | there are a lot of things i dont like |
| 01:31.20 | Milenco | need to figure out a way of distributing the configs thru puppet in a nice way |
| 01:31.31 | Milenco | because most of the code is the same on most asterisk instances |
| 01:32.36 | WIMPy | Yah, I was thinking of doing somethign to automate recycling. I was pondering using git. |
| 01:33.06 | WIMPy | Archive and distribute in one. |
| 01:33.19 | Milenco | with git autodeploy? |
| 01:33.44 | WIMPy | Not sure, yet. |
| 01:33.50 | Milenco | we maintain our code with git, but for these things its all stored in the puppet project |
| 01:34.10 | Milenco | puppet can reload/restart asterisk if needed |
| 01:34.21 | Milenco | not sure about git autodeploy |
| 01:34.48 | WIMPy | That doesn't really matter. That's easy enough to script. |
| 01:35.17 | WIMPy | My config generator automatically reloads what it has changed anyway. Nothign fancy, really. |
| 01:35.25 | Milenco | thats true. we prefer puppet as distribution because its made for it, git is made for version control |
| 01:35.33 | Milenco | we use what is best in what it does |
| 01:36.17 | WIMPy | Yes, but being able to revert a change possibly on a local level can be helpful. |
| 01:37.01 | WIMPy | And you could even manage local extras with it. |
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| 01:55.08 | Milenco | yes thats very true :) |
| 01:58.03 | WIMPy | It really depends on how much time I find to do it. I might just end up with a script that just does rsync;asterisk -x reload. |
| 02:03.23 | Milenco | i dont have much time usually :( |
| 02:03.37 | Milenco | thats why i want to setup it up in the most automated way :) |
| 02:03.45 | Milenco | anyway..its getting pretty late here |
| 02:03.48 | WIMPy | The most common issue, surely. |
| 02:03.50 | Milenco | going try and find my bed :) |
| 02:04.00 | Milenco | nice chatting! :) |
| 02:04.01 | WIMPy | Good luck :-) |
| 02:04.09 | Milenco | Cya! :) |
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| 03:06.21 | fling | Are you guys using Medooze MCU? |
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| 03:06.52 | fling | is paying $10 to anyone who will help to setup it. |
| 03:06.53 | WIMPy | What is it? |
| 03:07.12 | *** join/#asterisk saint_ (~saint@209.188.5.226) |
| 03:07.30 | fling | WIMPy: conference bridge replacement with video mixing support. |
| 03:07.58 | WIMPy | nice |
| 03:09.22 | saint_ | hi all - what would cause asterisk 13 not to send ring back to the caller ? |
| 03:09.32 | saint_ | i'm using answer(), dial(), and hangup . |
| 03:09.54 | saint_ | i have 2 way audio if i pick up. it's just the ringback tone which is missing on the caller side. |
| 03:10.02 | fling | saint_: try asterisk -rvvvv |
| 03:10.02 | WIMPy | You probably shouldn't use Answer(). |
| 03:10.12 | fling | saint_: paste what it is doing ;> |
| 03:10.21 | saint_ | WIMPy that s what i have in 1.8 ... i m trying to port the conf to 13 |
| 03:10.22 | WIMPy | Does Asterisk know something is ringing? |
| 03:10.53 | WIMPy | Answer() befor Dial() is usually bad. |
| 03:11.05 | saint_ | WIMPy why is that ? |
| 03:11.31 | WIMPy | Why do you want to answer a call that actually hasn't been answered? |
| 03:11.54 | saint_ | WIMPy i think i had to do it in 1.8 because i had a series of test (testing incoming number) to do before dialing the right person |
| 03:11.58 | Penguin | $10? What do you want me to do after I'm finished with the first 6 minutes you've paid for? |
| 03:12.36 | fling | Penguin: idk? Maybe someone will just provide a precompiled qemu/lxc/another image with everything set up. |
| 03:12.53 | WIMPy | saint_: That does not sound like a sensible reason. Things like Proceeding(), Ringing() or Progress() may be suitable. |
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| 03:13.20 | fling | Penguin: are you using Medooze? :P |
| 03:13.25 | saint_ | WIMPy nah.. i know i had a reason, because without it, whatever i tried to do in 1.8 was not working .. |
| 03:13.55 | WIMPy | saint_: There mus be something else wrong then. |
| 03:14.27 | saint_ | fling : http://pastebin.com/VGv3fLgT |
| 03:14.37 | saint_ | that's with vvvv , but it does not show anything relevamt |
| 03:17.05 | saint_ | WIMPy i guess it's because I had soem gotoif ... |
| 03:17.16 | saint_ | WIMPy and Set() .. |
| 03:17.34 | WIMPy | Then you should do it at the target that doesn't Dial(). |
| 03:17.56 | saint_ | WIMPy and also because I used to have google talk |
| 03:17.58 | WIMPy | Otherwise the log looks fine to me. |
| 03:18.07 | saint_ | WIMPy where I had to answer, and send a DTMF before ringing the call |
| 03:18.19 | saint_ | WIMPy so what would prevent a ring back tone ? |
| 03:18.24 | WIMPy | That's another building site. |
| 03:18.26 | saint_ | I can try without answer .. for now .. |
| 03:19.12 | WIMPy | As it has to be sent inband the only idea I have that there's something going wrong inside Asterisk. |
| 03:19.49 | saint_ | WIMPy - My apologies . |
| 03:19.53 | WIMPy | OTOH missing ringback seems to happen too often anyway. |
| 03:20.01 | saint_ | WIMPy removing the Answer makes the ringback tone work .. |
| 03:20.34 | saint_ | MMhh.. well.. I'm just trying the basics of Asterisk 13 right now. pjsip is a real pain in the butt, comparing to sip.conf |
| 03:20.50 | WIMPy | Ok, so it works if you do it the nice way. But it should also work the other way. Looks like a bug to me. |
| 03:20.56 | saint_ | i m just trying to have the basics work, before i import my extension.conf |
| 03:22.54 | fling | Penguin: no? :D |
| 03:24.23 | saint_ | have you guys tried Release 12 or 13, with pjsip ? |
| 03:24.40 | WIMPy | hasn't |
| 03:35.00 | saint_ | is there ANY clear documentation about pjsip that would break down the sections, and the possible values, beside this not-so-friendly doc ? https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_aors |
| 03:35.31 | [TK]D-Fender | The sample config |
| 03:36.18 | saint_ | [TK]D-Fender i ve been using it. i was looking for better than that :D |
| 03:39.07 | saint_ | what does aor stand for ? |
| 03:41.51 | [TK]D-Fender | Authority Of Record |
| 03:42.12 | saint_ | thanks |
| 03:44.15 | [TK]D-Fender | Address Of Record rather |
| 03:44.20 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Sections+and+Relationships |
| 03:46.50 | saint_ | Thank you [TK]D-Fender |
| 03:47.59 | saint_ | I was reading this page instead: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_aors |
| 03:57.18 | [TK]D-Fender | Want generic info aim for the general docs, not something with a version # attached to it |
| 03:57.18 | mjordan | saint_: That's the reference page. It's just a listing of all possible options. There's a lot of examples for actually configuring this in the Configuration section. |
| 03:57.56 | mjordan | https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples |
| 03:58.15 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-plcxfkozgsplmvap) |
| 04:08.19 | saint_ | going to bed . ttyl8r |
| 04:09.05 | ChannelZ | tty closed |
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| 06:16.39 | _omer | is there any special instruction to install asterisk as non root? |
| 06:18.56 | gryphon | _omer, install or run as non root? |
| 06:20.08 | _omer | install + run |
| 06:21.50 | gryphon | cant say anything about install, but to run as non root you only need to set options 'runuser' and 'rungroup' in asterisk.conf |
| 06:21.58 | _omer | what is the difference between ...... a) if asterisk is installed as root but running as non-root ...... b) asterisk was installed as non-root and also running as non-root? |
| 06:24.06 | _omer | gryphon: if I change 'runuser' and 'rungroup' in asterisk.conf then do I need to change permissions of /etc/asterisk from root to the non-root user? |
| 06:25.32 | gryphon | non-root user need access to read configs |
| 06:27.55 | _omer | ok. thanks |
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| 06:35.55 | fling | I still need help with Medooze MCU. |
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| 07:47.28 | ChannelZ | gryphon: the asterisk user also needs write access to several directories, like voicemail, the AstDB, logs.. |
| 07:57.42 | gryphon | ChannelZ, yes, of course, but question was about etc ) |
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| 11:15.08 | Geek-Linux | hello all: can i configure asterisk to send another RTP ip address in the 200 OK packet, other than the IP address on with INVITE packet is received ? |
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| 11:32.15 | kaldemar | Geek-Linux: is your asterisk behind a NAT? |
| 11:33.18 | Geek-Linux | kaldemar: no |
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| 13:21.26 | jkroon | hi guys, i've received a bunch of files on disk that contains * chars in the filenames, trying to use asterisk -rnx "file convert ... ..." to convert those files but asterisk keeps failing - how can I escape the filename such that asterisk will absorb and convert it? |
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| 13:25.23 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:25.58 | jkroon | nm, it's not an asterisk issue ... g729 license file borked again. |
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| 14:41.11 | n3tctrl | I am in need of a recommendation for an external T1 pri line gateway. anybody have a preferernce or experience with any that i should avoid? |
| 14:42.12 | WIMPy | Are you sure you want it external? |
| 14:43.23 | n3tctrl | <WIMPy> yes the dmarc and server will be located almost 3,000 feet apart on the other side of our facility. we are consolidating our employees from scattered amongst 5 buildings to a single building. we have copper to most of it, but we have fiber everywhere. |
| 14:43.45 | n3tctrl | my plan is to convert to IP at the dmarc, and run ip to the other side of campus. |
| 14:46.06 | newtonr | n3tctrl, I'm biased as I work for Digium : http://www.digium.com/en/products/voip-gateways |
| 14:46.41 | newtonr | Supporting Digium supports the Asterisk project :D that is all the marketing I'll do today - and I'm out be I'll be back later! |
| 14:46.43 | WIMPy | Which is as good as putting an Asterisk there. |
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| 15:16.23 | Katty | anyone want to exchange xmas cards with me this year? |
| 15:18.30 | PhirePhly | aw crap. I need to start putting mine together |
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| 15:36.48 | Katty | pokes [TK]D-Fender |
| 15:44.09 | [TK]D-Fender | Katty: Sentiment alone is nice, no need for postage :) |
| 15:52.33 | Katty | very well, dear. |
| 15:52.35 | Katty | hugs [TK]D-Fender |
| 15:52.47 | Katty | have you been doing alright? |
| 15:53.33 | [TK]D-Fender | Meh in general, but lifestyle improving... |
| 15:54.13 | Katty | well improving is always good (= |
| 15:56.33 | WIMPy | wonders if the special drug consumption area opens today or tomorrow. |
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| 16:19.50 | Mango45 | Is it possible to configure a jitter buffer for a specific peer(s)? The only way I've figured out how to do it so far is in the [general] section of sip.conf. |
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| 18:23.11 | lorsungcu | not really asterisk related, but Iâm having issues with digium.updateXmlConfig() properly rebooting/reconfiguring a D40/45, anyone have experience troubleshooting this? |
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| 19:56.21 | tuc0 | hey guys, is there a way to pass a variable between two legs of a channel? |
| 19:56.58 | WIMPy | Hmm. What was it called? MASTER_CHANNEL? |
| 19:57.03 | tuc0 | ie, you use AMI to originate a call to a local context and both legs are running through the dial plan but you'd like to pass a variable between them |
| 19:57.52 | tuc0 | thanks WIMPy, I'm looking into it. |
| 19:58.00 | WIMPy | Not sure what the added local channels would do to that idea. |
| 19:58.12 | tuc0 | I'll find out. |
| 20:00.08 | babak | Hi,for writing a dialer via AGI, is there a way to write it asynchronously(non-blocking) ? |
| 20:00.34 | WIMPy | babak: What are you trying to do? |
| 20:00.56 | tuc0 | babak, someone correct me if I'm wrong, but if you're just trying to originate calls you should probably use AMI not AGI |
| 20:01.16 | babak | make a very big voice broadcasting with many Asterisk boxes |
| 20:01.22 | WIMPy | It's more like you can't do that with AGI. |
| 20:01.49 | tuc0 | WIMPy, well, you *could* do it with AGI, painfully. |
| 20:02.10 | WIMPy | Only if you already have a call to run that AGI on. |
| 20:02.17 | tuc0 | setup call files to hit the dialplan to initiate it |
| 20:02.20 | tuc0 | or ami |
| 20:02.23 | tuc0 | right |
| 20:02.39 | WIMPy | There are 3 ways to originate calls: CLI, call files an AMI. |
| 20:02.53 | tuc0 | I didn't realize you could originate off the CLI |
| 20:03.03 | tuc0 | that could have saved me a lot of time in some cases for testing |
| 20:03.10 | WIMPy | channel originate |
| 20:03.15 | tuc0 | thanks |
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| 20:03.19 | WIMPy | What do yu do for testing? |
| 20:03.46 | tuc0 | either a call file, ami, or I have a SIP phone sitting here |
| 20:04.08 | babak | with AMI is it possible to play voice broadcast and meanwhile process DTMF input and transfer call to some IVRs like survey |
| 20:04.09 | babak | ? |
| 20:04.34 | WIMPy | babak: No. That are two different things. |
| 20:05.00 | tuc0 | use AMI to originate the call and the call connects to wherever you specific in the dialplan. Then you make the dialplan do all that fancy footwork. |
| 20:05.00 | WIMPy | First you originate a call and the you send it somewhere, like an extension in your dialplan. |
| 20:05.10 | tuc0 | Or you could have the dialplan run an AGI script/program |
| 20:07.04 | tuc0 | babak, I would suggest learning how to build simple asterisk dialplans. Everything you require can be done rather easily without having to write some AGI script for it. |
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| 20:09.46 | babak | tuc0: is it possible to initiate many many calls with dialplan? I need to use AMI for Asynchronous(non-blocking) feature and scalable for distributing calls between many boxes |
| 20:10.15 | WIMPy | You don't initiate calls with dialplan. |
| 20:10.19 | tuc0 | you originate many many calls with AMI using basic sockets |
| 20:10.48 | WIMPy | You can usr all 3 ways to originate calls in parallel. |
| 20:10.59 | tuc0 | write a program that is multithreaded and can connect to multiple AMI sockets, port 5038 default, send em out as needed to your different boxes all running the same dialplan |
| 20:11.17 | tuc0 | keep track of which is handling what and load balance |
| 20:11.29 | WIMPy | Err. You neither need multiple socktes nor multithreading. |
| 20:11.51 | WIMPy | Oh, for the different boxes. ok, but still no need for multithreading. |
| 20:11.54 | tuc0 | well otherwise you're going to connect and authenticate to each one whenever you want to originate a call and you lose all of the event data? |
| 20:12.05 | tuc0 | including when the call has finished so you know it's current load |
| 20:14.23 | tuc0 | I mean you could run a select on the socket handles and do it that way of course if threading is too much work, but I wouldn't want to risk a buffer being over filled while I'm processing a different socket |
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| 20:19.18 | babak | thx |
| 20:19.22 | tuc0 | WIMPy, the channel originate command is awesome, I can't believe I didn't know about it. Thanks again. |
| 20:19.41 | WIMPy | RTFM *eg* |
| 20:20.03 | WIMPy | Far too many options to remember them all. |
| 20:21.41 | tuc0 | hmm? |
| 20:22.29 | WIMPy | The Asterisk manual is just too long to know everything. |
| 20:24.39 | tuc0 | Oh right. Yeah learning something new every day, I started 10 years ago when asterisk version <1.0 for a few years and just started back up. A lot of new fun stuff. I've been trying to wrap my head around ARI for instance, but I'm still not exactly sure how I would use it. |
| 20:30.59 | tuc0 | Does anyone paying attention in here have any applications for ARI? I just want an example of why you would ever use it over what already exists. |
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| 20:57.09 | newtonr | tuc0, Read through https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573 |
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| 21:02.43 | saint_ | is the acl.conf in asterisk 13 valid for connection coming from the sip trunk (like a SIP INVITE) , or for end points to register ... or both ? |
| 21:04.15 | newtonr | I believe it depends on the module using the ACL, ping file putnopvut |
| 21:04.52 | mjordan | actually, putnopvut had little to do with that particular thing |
| 21:04.53 | mjordan | :-) |
| 21:05.00 | file | or me |
| 21:05.15 | mjordan | As it says at the top of the sample config, acl.conf is used to define shared ACLs that you can refer to by name elsewhere |
| 21:05.42 | mjordan | if an 'item' - in sip.conf. pjsip.conf, iax.conf, etc. - allows for the usage of named ACLs, then it can reference the ACL defined in acl.conf |
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| 21:06.45 | saint_ | ok, thanks |
| 21:06.50 | newtonr | file, I just figured you may know. You know things and stuff. :) |
| 21:10.34 | file | I know NOTHING |
| 21:12.01 | newtonr | saint_, are you using pjsip in Asterisk 13? If you define an ACL there (or call one from acl.conf), I believe it applies only to permitting or denying inbound requests. |
| 21:12.27 | newtonr | Hey my documentation says |
| 21:12.29 | newtonr | "operates on all inbound SIP communication using res_pjsip." |
| 21:12.36 | newtonr | yay |
| 21:19.30 | tuc0 | newtonr, I have read that, perhaps I'm just dense but I don't really see a need for it. It's not a replacement for AMI or AGI, it's only purpose is for writing applications in a language other than C, is that right? |
| 21:20.33 | tuc0 | I also don't understand the purpose of using websockets and REST. why not just a TCP socket? |
| 21:20.34 | WIMPy | The thing that bothers me is that it again uses two different communication channels, just as you would when combining (fast)AGI and AMI. |
| 21:20.46 | WIMPy | But it definitely scores on the buzzword bingo. |
| 21:21.08 | tuc0 | WIMPy, so I'm right, that's all it is? Does it use the cloud? |
| 21:21.31 | WIMPy | That word is missing, I think. |
| 21:22.40 | file | it could very easily have it go all over websockets, but noone has written that yet |
| 21:23.39 | tuc0 | Well if they do that I hope they put in a basic TCP interface to it as well, I'm not really sure I understand the need for websockets |
| 21:23.54 | tuc0 | Because a single channel for all of this would be awesome. |
| 21:24.22 | WIMPy | We already have AMI. It would just need a little improvement. |
| 21:26.23 | tuc0 | You're right but I feel like AMI isn't going to see much improvement (I hope I'm wrong). So if we're forced to "upgrade" to ARI to get the same features I'm just hoping they don't force us to use websockets/rest exclusively. |
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| 21:27.31 | [TK]D-Fender | heads home |
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| 21:54.59 | pjensen00 | I have a truly bizarre scenario. I'm using the AGI command get_data in many places. It used to work just fine. However, In my AGI script I"m doing a rest request to pull some external data (nothing directly asterisk related). However, the next get_data I do after that rest request automatically gives me a return value of '6' instead of playing the message. |
| 21:55.14 | pjensen00 | Then the next get_data and beyond functions normally |
| 21:55.56 | pjensen00 | I'm completely baffled how doing a REST request would have anything to do with the DTMF detection of asterisk. |
| 21:57.58 | pjensen00 | Anyone have any clue as to what I could possibly be doing? I've done the get_data on the line before and after the REST request multiple times to make sure that it defintely is the trigger point that puts my AGI script in the weird state. |
| 21:58.24 | WIMPy | Are you using some library or did you write the stuff yourself? |
| 22:00.10 | pjensen00 | Using the LWP library in Perl |
| 22:00.59 | pjensen00 | for the request, using an AGI library Asterisk::FastAGI for the agi library |
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| 22:04.58 | babak | tuc0: it seems another way for writing scalable voice broadcast system is ARI |
| 22:05.30 | babak | it is asynchronous |
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| 22:23.12 | pjensen00 | OK. So this is fascinating, the reason why the get_data bit is freaking out is because the channel status is 0 or blank. Doing the get_data resets the channel status 6 (answered) |
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| 22:34.54 | tuc0 | babak, you don't want to use ARI for originating calls, it's meant to replace applications. Use AMI, it is asynchronous just like you require. |
| 22:35.14 | tuc0 | just add async: true to your originate commands |
| 22:37.31 | tuc0 | pjensen00, are you sure you don't have a memory leak of some kind? |
| 22:39.19 | file | you could use ARI to originate calls, nothing to say you can't |
| 22:40.56 | tuc0 | file, I thought it wasn't an AGI or AMI replacement in that it couldn't do what they could. Is there any documentation on originating a call with ARI? |
| 22:41.03 | tuc0 | I couldn't find anything the other day when I looked |
| 22:42.19 | file | I don't think there's a howto specifically... |
| 22:42.40 | file | it's done by doing an HTTP POST to the channels resource, you tell it the dial string and where you want it to go (dialplan location or Stasis application) |
| 22:43.41 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Channels+REST+API#Asterisk13ChannelsRESTAPI-originate |
| 22:44.31 | tuc0 | file, cool thank you |
| 22:44.38 | pjensen00 | tuc0: I don't know. I use this library in other non asterisk related areas and it doesn't appear to cause any kind of side effects |
| 22:48.03 | tuc0 | pjensen00, you could do a quick test by sending the command manually and skip the library. Or load up asterisk with -vvvvc directly and watch the data get sent/received on std err. assuming your library does that. |
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| 22:48.52 | pjensen00 | I wonder if that'll conflict with the logger I use that redirects STDOUT/STDERR to a log file. |
| 22:52.45 | tuc0 | well just check your log file, I mean if result=0 or result=1 (0) is returned then you know asterisk is probably the guy screwing up. |
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| 22:56.50 | pjensen00 | thanks! |
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