| 00:01.06 | *** join/#asterisk barbosa2 (~juliano.b@177.158.4.82) |
| 00:13.00 | *** join/#asterisk BlackDex (~BlackDex@ori.vyus.nl) |
| 00:16.50 | *** join/#asterisk felipealmeida (~user@177.41.19.159) |
| 00:46.39 | *** join/#asterisk suneye (~atcmmi@119.122.153.183) |
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| 01:18.31 | *** join/#asterisk xzarth_ (~krikkit@dh207-25-208.xnet.hr) |
| 01:20.17 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 01:20.33 | snadge | time test.. i googled it.. failed.. i must suck at googling ;) |
| 01:21.40 | snadge | SayUnixTime sounds like it could be it |
| 01:23.16 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-zruxehlkxnaczthc) |
| 01:23.16 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:24.18 | mjordan | Qwell: you around? |
| 01:24.39 | Penguin | snadge: What was the question? |
| 01:24.41 | snadge | my extensions.conf has both a [time] and an [ani] in them.. how do i use them? |
| 01:25.05 | snadge | I just want to set an extension.. say.. 1234 to answer and say the time.. or some interesting information |
| 01:25.15 | Penguin | Those are just contexts. Contexts have extensions in them. |
| 01:25.55 | snadge | right so they're set to _X instead of 1234.. or something like that |
| 01:26.25 | Penguin | That sounds like a terrible pattern. |
| 01:26.44 | snadge | presumably i can say something like .. exten => 1234,1,ani |
| 01:27.06 | snadge | im a total asterisk noob.. but that's okay .. i'm sure theres some books and things I can go off and read |
| 01:29.23 | mjordan | ~book |
| 01:29.24 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 01:30.14 | snadge | that might not be the most efficient way to set up an extension.. which simply echos the time |
| 01:30.48 | snadge | but i'll order the book anyway |
| 01:31.00 | snadge | ~buybook |
| 01:31.00 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/0636920025894 so go buy it SERIOUSLY |
| 01:31.22 | snadge | what version does the book cover? |
| 01:33.05 | mjordan | snadge: versions up to 11 |
| 01:34.18 | Penguin | snadge: http://pastebin.com/kNuiSn03 |
| 01:39.08 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 01:39.37 | SeRi | Penguin: is your test chan off line? |
| 01:40.08 | Penguin | I doubt it, considering my asterisk is never off. |
| 01:40.17 | Penguin | Are you trying it by IP address? |
| 01:40.42 | SeRi | Penguin: let me see |
| 01:40.59 | SeRi | yes by IP |
| 01:41.04 | SeRi | I cant get in |
| 01:41.09 | SeRi | I am testing my system |
| 01:41.30 | Penguin | I probably have a different address since you last tried. |
| 01:41.36 | SeRi | ah ok. |
| 01:41.53 | Penguin | I'll PM you the current IP address. |
| 01:42.10 | SeRi | can you msg me the new IP if you don't mind? I am testing a possible issue on my sys |
| 01:44.13 | SeRi | Penguin: I am done. |
| 01:44.14 | SeRi | Thanks |
| 01:44.20 | SeRi | It's not my issue |
| 01:44.28 | SeRi | it's dijib's busted shit |
| 01:44.33 | Penguin | I saw you came and went. |
| 01:44.35 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 01:44.41 | SeRi | Penguin: yup |
| 01:46.54 | SeRi | dijib's shit is allways broken |
| 01:47.37 | Penguin | Yep. |
| 01:47.55 | SeRi | lol |
| 01:47.56 | Penguin | That happens when you don't have any idea what you're doing. |
| 01:48.08 | SeRi | rofl |
| 01:48.47 | Penguin | If you copy and paste from someone else's broken shit, guess what you'll end up with! |
| 01:48.54 | *** join/#asterisk dijib (~dijib@24-231-75-151.eastlink.ca) |
| 01:49.08 | Penguin | And that was how I hacked dijib's asterisk box. |
| 01:49.15 | SeRi | bad ass |
| 01:49.26 | Penguin | Oh, look. dijib. |
| 01:49.32 | dijib | lol |
| 01:49.48 | SeRi | I am going to try it. Penguin |
| 01:49.52 | SeRi | o wait I got root |
| 01:49.56 | SeRi | nice |
| 01:49.58 | Penguin | haha |
| 01:50.00 | dijib | i really dont understand what you were saying and will have to look at the logs |
| 01:50.19 | snadge | wow i suck at asterisk.. i've defined a 777 extension.. now i get an error Call from '' (192.168.1.120:5060) to extension '777' rejected because extension not found in context 'public'. |
| 01:50.34 | Penguin | dijib: If you'd take the time to LEARN how asterisk does things, you'd have less trouble. |
| 01:51.02 | Penguin | snadge: Is 777 in public? |
| 01:51.17 | SeRi | Penguin: lol! |
| 01:51.28 | SeRi | Penguin: +1 |
| 01:51.31 | snadge | oh man.. why did i even get out of bed today |
| 01:51.42 | snadge | public includes demo.. so it has to be within demo context or in there to work |
| 01:51.44 | snadge | *facepalm* |
| 01:52.17 | SeRi | dijib: almost 3yrs now. shame. :/ |
| 01:52.29 | dijib | why shame? my shit works, blame digium |
| 01:52.46 | SeRi | blame the op. o hold on. user. |
| 01:53.48 | SeRi | drop it like is hot |
| 01:54.17 | SeRi | dijib: tron can help you. msg him. |
| 01:54.37 | Penguin | It's really rather simple. |
| 01:54.55 | dijib | Asterisk 11.6.0 built by root @ swisscomm on a i686 running Linux on 2013-10-29 01:46:29 UTC |
| 01:55.40 | dijib | Penguin: conf |
| 01:55.44 | dijib | 2663 |
| 01:55.46 | Penguin | YAY! |
| 01:55.46 | dijib | at my domain |
| 01:55.56 | SeRi | if it does not work recompile. |
| 01:56.04 | SeRi | lol the dijib way |
| 01:56.10 | Penguin | It's most likely a configuration error. |
| 01:56.42 | dijib | dude ive recompiled and it works,. didnt reconfigure |
| 01:56.52 | snadge | Gosub address is invalid: 'time' |
| 01:57.00 | Penguin | Why would you need to gosub? |
| 01:57.04 | SeRi | dijib: so what was broken? |
| 01:57.16 | dijib | your face |
| 01:57.20 | SeRi | lol |
| 01:57.22 | Penguin | That is for a sub-routine, which will return back to where you came from when it is done. |
| 01:57.26 | SeRi | I rest my case. |
| 01:58.08 | Penguin | You can either change exten 'time' to a number, or use a Goto(). |
| 02:00.43 | dijib | Penguin: did u ever get that sangoma card operational? |
| 02:00.51 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-zruxehlkxnaczthc) |
| 02:00.53 | Penguin | No. |
| 02:00.58 | dijib | damn i want that card |
| 02:01.02 | dijib | could use |
| 02:01.07 | Penguin | I've never had a computer with the right slot in it. |
| 02:01.29 | dijib | 3 seconds too long |
| 02:01.34 | dijib | maybe 2 |
| 02:01.49 | dijib | thanks fior the tt-monkeys in our ears |
| 02:02.22 | dijib | Penguin: |
| 02:03.10 | SeRi | LOL |
| 02:03.11 | dijib | stop being a dick and join |
| 02:03.25 | SeRi | LMAO |
| 02:03.29 | dijib | if thats a test, yes tt-monkeys plays ..... forever |
| 02:03.56 | SeRi | Penguin: The hack you show me is awesome! |
| 02:04.01 | dijib | Executing [2663@conference:9] ConfBridge("SIP/anonymous.invalid-00000008", "2663,norec_bridge,default_user,user_menu") in new stack |
| 02:04.06 | dijib | "Hacking" |
| 02:04.09 | Penguin | seri: oh. |
| 02:04.19 | SeRi | Penguin: LOL |
| 02:05.38 | SeRi | flooood!!!! |
| 02:05.50 | SeRi | monkeys! |
| 02:06.12 | *** join/#asterisk zendel (~zendel.fe@chatswood.au.ulterius.net) |
| 02:07.29 | dijib | do it again |
| 02:07.33 | dijib | its fun |
| 02:07.38 | SeRi | yea Penguin do it again |
| 02:07.38 | zendel | hi!, IAX2 show netstats shows the below values, |
| 02:07.52 | Penguin | Do what again? |
| 02:09.03 | dijib | flood my shit |
| 02:09.09 | Penguin | UDP? |
| 02:09.13 | dijib | SIP |
| 02:09.18 | Penguin | No can do. |
| 02:09.23 | zendel | hi!, IAX2 show netstats shows the below values, |
| 02:09.23 | zendel | RTT Jit Del Lost |
| 02:09.23 | zendel | 625 -1 0 -1 |
| 02:09.23 | zendel | Can anybody share some thoughts regarding the -1 values |
| 02:09.36 | dijib | chan orig application playback |
| 02:09.38 | dijib | fun |
| 02:10.20 | SeRi | LOL |
| 02:10.22 | SeRi | ROFL |
| 02:11.12 | SeRi | I have a conffesion to make dijib. It's not Penguin. look closer at your logs. |
| 02:11.56 | Penguin | I didn't do anything, but I'm considering installing sipvicious just for dijib. |
| 02:12.04 | SeRi | LOL |
| 02:12.26 | SeRi | its not to hard to get to his system actually |
| 02:13.08 | Penguin | I'm not all that concerned with it at this point. |
| 02:13.08 | dijib | k stop |
| 02:13.17 | dijib | this is getting annoying and i cant find your IP to block |
| 02:13.23 | Penguin | haha |
| 02:13.44 | Penguin | Step one: become a sysadmin |
| 02:13.44 | SeRi | lol to funny. |
| 02:13.50 | dijib | im going to cry |
| 02:14.01 | Penguin | Step two: find IP address |
| 02:14.02 | dijib | step one, lr2n cmd |
| 02:14.18 | Penguin | Step three: learn iptables |
| 02:14.25 | SeRi | rofl |
| 02:14.33 | Penguin | Step four: |
| 02:14.35 | Penguin | Step five: profit |
| 02:15.06 | dijib | step 5 you no profit cuz sangoma is still in drawer |
| 02:15.07 | SeRi | w0rd |
| 02:15.18 | Penguin | That's a true story. |
| 02:15.25 | Penguin | It's right where I put it several months ago. |
| 02:15.38 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 02:15.43 | SeRi | lol |
| 02:15.48 | Penguin | snadge: You missed some stuff. |
| 02:15.52 | dijib | get POTS already dude |
| 02:15.58 | dijib | sell it |
| 02:16.01 | SeRi | Talking about that... did you ever used the headset? |
| 02:16.02 | snadge | i did .. my bouncer failed.. spectacularly |
| 02:16.02 | dijib | something |
| 02:16.03 | Penguin | Why? I have the blagoblag. |
| 02:16.11 | snadge | i have forced the server to an australian one now.. maybe that will help |
| 02:16.18 | dijib | wtf is a blagoblag? |
| 02:16.18 | Penguin | (2057.00) <Penguin> Why would you need to gosub? |
| 02:16.18 | Penguin | (2057.22) <Penguin> That is for a sub-routine, which will return back to where you came from when it is done. |
| 02:16.21 | Penguin | (2058.07) <Penguin> You can either change exten 'time' to a number, or use a Goto(). |
| 02:16.30 | SeRi | w0rd |
| 02:16.58 | snadge | well.. my understanding of gosub and goto.. is it should make no difference.. except for obviously returning instead of not |
| 02:17.26 | snadge | unless it doesnt include a return.. and thats an error |
| 02:17.56 | Penguin | But do you need to return from the time just to hangup? |
| 02:20.08 | Penguin | dijib: Never heard of that before? |
| 02:20.10 | Penguin | http://xkcd.com/181/ |
| 02:20.11 | snadge | yeah.. the error is now.. No such label 'time' in extension '777' in context 'public' |
| 02:20.27 | snadge | thats because [time] is its own extension .. not inside public |
| 02:20.31 | Penguin | snadge: Pastebin the dialplan and I'll explain what's wrong. |
| 02:20.40 | Penguin | [time] is a context, not an extension. |
| 02:20.48 | Penguin | exten => time is an extension. |
| 02:20.59 | snadge | yes thats what i meant.. im easily confused :P |
| 02:21.36 | Penguin | But you're calling extension '777' in context 'public' looking for a label 'time' |
| 02:21.36 | snadge | so pastebin all of extensions.conf ? or just my 777 extension |
| 02:21.36 | *** join/#asterisk atcmmi (suneye@116.25.194.0) |
| 02:21.42 | Penguin | Just the one you're having problems with for now. |
| 02:21.48 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 02:21.48 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:22.18 | snadge | i've literally just put "exten => 777,1,Goto(time)" |
| 02:22.27 | snadge | in the [demo] context.. which is included from public |
| 02:22.51 | Penguin | That makes sense, then. Goto(time) means go to extension '777' label 'time'. |
| 02:23.06 | Penguin | But you a label in extension 777 that is 'time'. |
| 02:23.11 | Penguin | whoa... |
| 02:23.18 | Penguin | But you don't have a label in extension 777 that is 'time'. |
| 02:23.47 | Penguin | Brain hiccup, missed some words. |
| 02:24.04 | snadge | right.. im also quite confused why i haven't been able to find a trivial example of how to create an extension which echos the time ;) |
| 02:24.14 | snadge | the learning process is helpful though im sure |
| 02:24.21 | Penguin | I already gave you the whole thing! |
| 02:25.00 | snadge | ok.. so if i dont have a label in extension 777 that is time.. then im obviously doing something wrong |
| 02:25.07 | Penguin | Exactly. |
| 02:25.28 | snadge | i have to define one? or do it a different way |
| 02:25.31 | Penguin | A label 'time' would look like this: same => n(time),Stuff() |
| 02:25.59 | Penguin | or: exten => 777,n(time),Stuff() |
| 02:26.20 | Penguin | You don't need a time label, you need a time extension. |
| 02:26.34 | Penguin | Or just don't use extension 'time' but use a number instead. |
| 02:27.47 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:31.43 | snadge | lol thank you for your patience.. again, I just realised i didn't read the example properly |
| 02:31.49 | snadge | which has "include => time" |
| 02:32.09 | snadge | so now the example is working.. but its missing audio files for reading digits etc |
| 02:32.09 | Penguin | That would include the time CONTEXT. Is that what you want to do? |
| 02:32.21 | snadge | well now the goto/gosub works |
| 02:32.38 | Penguin | Not what you showed me. |
| 02:34.21 | Penguin | You first need to know what a context is, what an extension is, what priorities are, and what priority labels are. |
| 02:35.56 | snadge | i do need to know that :) |
| 02:36.07 | Penguin | Start with context. |
| 02:36.19 | Penguin | Contexts look like [this] |
| 02:36.47 | Penguin | Contexts are logical containers for extensions. They are used to group extensions. |
| 02:37.41 | Penguin | Once you create a context, you can then create one or more extensions within the context. |
| 02:38.23 | snadge | got it.. logical |
| 02:38.33 | Penguin | The context starts with the name (which is in square brackets) and goes until the next context name (which is in square brackets). |
| 02:39.33 | snadge | unrelated.. it seems my 777 extension is not inheriting the default language context |
| 02:39.36 | Penguin | You can't have the same extension duplicated within the same context, but you can have a duplicated extension within another context. Remember, containers for extensions. |
| 02:40.06 | snadge | got it.. and obviously if you include a context which has a duplicate extension in it.. it will error ? |
| 02:40.15 | snadge | or it will override it |
| 02:41.13 | Penguin | It will throw out a warning that you have duplicated the extension. |
| 02:41.14 | snadge | so my sip.conf has "language=en_AU" in it.. i've checked that the audio files for digits/oh exist .. in en_AU |
| 02:41.44 | snadge | so something else must be overriding the language back to default (US) im assuming |
| 02:42.18 | Penguin | core show settings |
| 02:42.24 | Penguin | Look at Default language. Is it showing what you expect? |
| 02:42.51 | Penguin | Does it say the Language prefix is enabled or disabled? |
| 02:42.58 | snadge | hmm.. it says default language is en |
| 02:43.14 | snadge | language prefix is enabled |
| 02:43.25 | Penguin | Did you run sip reload after setting the language in sip.conf? |
| 02:43.41 | snadge | i haven't changed that setting in a long time.. and have restarted the server several times |
| 02:43.46 | Penguin | okay |
| 02:44.12 | Penguin | Oh, I see the discrepancy... |
| 02:44.19 | Penguin | asterisk.conf |
| 02:44.20 | Penguin | defaultlanguage |
| 02:44.23 | snadge | which is odd.. for the echo test, and the IAX digium test.. it selects the right language audio |
| 02:44.48 | Penguin | I've never had a language problem, so I'm not sure what sets what and where. |
| 02:45.19 | snadge | yeah.. i'll make a note of the asterisk.conf setting.. i wonder why both settings exist, and what the difference is |
| 02:45.56 | Penguin | The one in asterisk.conf is going to be the core language. I had no idea what the sip.conf language setting will control. I use the default en everywhere. |
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| 03:17.45 | snadge | its working now :) |
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| 04:45.48 | snadge | so apparently the latest asterisk 1.8 contains a bunch of fixes for faxing |
| 04:45.59 | snadge | which is why faxing doesn't work anymore ;) |
| 04:46.21 | snadge | other sysadmin is looking at downgrading it to 1.8.15-cert3 |
| 04:46.37 | *** join/#asterisk charkee (~charkee@121.54.46.199) |
| 04:47.09 | snadge | i see.. its certified.. well that does sound exciting |
| 04:47.17 | Penguin | What's broken? |
| 04:47.59 | snadge | im the new guy here obviously.. but from my understanding, our upstream carrier changed something yesterday.. which broke dtmf detection |
| 04:48.07 | snadge | we were running some old version of 1.8 .. something like 1.8.9 |
| 04:48.35 | snadge | so in the interest of just getting things working again.. it was updated to the latest 1.8 .. 1.8.24.0 |
| 04:48.44 | snadge | and this fixed the dtmf issue.. happy days |
| 04:48.57 | Penguin | I was running 1.8.8.0 for a long time. I just recently upgraded to 1.8.23.1, but I haven't received any faxes since the upgrade. |
| 04:49.41 | snadge | apparently now faxes just arn't negotiating.. it might have something to do with fixing the input and output codecs to alaw |
| 04:50.09 | snadge | i sent a test fax earlier.. and you could hear the fax trying to negotiate a few times.. then the fax machine just says error |
| 04:50.20 | Penguin | You've got to use alaw, though. Other codecs aren't fax-compatible. |
| 04:50.33 | snadge | right |
| 04:50.47 | snadge | so i was starting to question the logic of why the other sysadmin wants to downgrade.. yet again |
| 04:50.56 | snadge | i wasn't aware of the certified versions |
| 04:51.58 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 04:52.31 | snadge | and i find it difficult to argue with that decision.. certified does sound good, for something that's basically running our upstream trunk |
| 04:53.44 | Penguin | I didn't see the certified releases until recently. I kind of feel like they are a gimmick. |
| 04:54.13 | snadge | well.. i would happen to agree, my familiarisation with other projects suggests that to likely be the case |
| 04:54.56 | Penguin | Who certifies it? What are the certification criteria? |
| 04:56.53 | snadge | i understand the logic behind it though.. the way the particular organisation that I work for uses asterisk, is quite complex |
| 04:57.18 | snadge | you have big asterisk servers that are connected to upstream providers.. which basically just act as a switch |
| 04:57.29 | Penguin | Does the certification process meet your use case? How about mine? And all the others, too? |
| 04:57.31 | snadge | and we have a pool of asterisk servers.. which handle calls |
| 04:57.55 | snadge | my basic understanding is that asterisk shouldn't be used as a switch.. but whatever.. its a termination point, not a router |
| 05:01.28 | Penguin | I'm off to bed. |
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| 06:29.17 | snadge | im just running a sip load tester on my netscaler / test environment |
| 06:29.38 | snadge | never mind i answered my own question ;) |
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| 06:30.21 | snadge | <PROTECTED> |
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| 06:45.05 | snadge | actually that didnt fix it.. one of my servers is configured to dial out our trunk |
| 06:46.12 | snadge | and it has a catch all: exten => _X.,1,Dial(SIP/ftel/${EXTEN}) |
| 06:46.31 | snadge | anyway.. when i try to dial my 777 extension on that server.. it goes out ftel |
| 06:46.36 | snadge | and of course fails |
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| 06:53.17 | snadge | i tried including the demo context from without that outgoing context |
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| 07:08.18 | gerritfromsa | anyone awake yet |
| 07:08.42 | *** join/#asterisk Tokeiito (~quassel@main.kbi.lt) |
| 07:08.43 | gerritfromsa | or not gone to bed ... |
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| 07:31.50 | the_5th_wheel | Has anyone had any issues with mixmonitor and the voice sounding choppy / robot like? I think its related to my calls running on g729 with a packetsize of sixty |
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| 07:45.30 | gerritfromsa | it will do this if your service provider used something else and asterisk must transcode between 60ms and 20ms for example |
| 07:46.09 | the_5th_wheel | Ive looked at the traffic dumps and both legs seem to be 60ms |
| 07:46.13 | the_5th_wheel | is there any way to get past this? |
| 07:46.46 | gerritfromsa | If its 60 all the way it should not do this |
| 07:47.08 | gerritfromsa | I has the same issue about a year ago and fixed it by making sure they're all the same |
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| 07:47.43 | gerritfromsa | you've got allow=g729:60 in there ? |
| 07:47.58 | gerritfromsa | Using open source g729 or licensd one? |
| 07:50.13 | the_5th_wheel | I was using the licenced one but out of desperation I tried the open source on too |
| 07:50.17 | the_5th_wheel | and made no difference |
| 07:50.30 | the_5th_wheel | and yes disallow=all allow=g729:60 |
| 07:51.39 | gerritfromsa | you double checked with wireshark that the delta between the packets are 60? in and out ? |
| 07:51.53 | the_5th_wheel | I did, but let me do a triple check :-) |
| 07:52.14 | gerritfromsa | nothing wrong with triple |
| 07:53.19 | *** join/#asterisk suneye (~atcmmi@119.122.153.183) |
| 07:53.40 | gerritfromsa | going to town ... later |
| 08:06.02 | the_5th_wheel | Triple checked. Def same packet size (both in the sip negotiation and the actual RTP packets) |
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| 09:20.40 | hos7ein | hi |
| 09:21.53 | hos7ein | i write this config on /etc/asterisk/sip.con |
| 09:21.54 | hos7ein | http://paste.fedoraproject.org/50117/38500138/ |
| 09:22.21 | hos7ein | and file /etc/asterisk/sip_additional.conf |
| 09:22.56 | hos7ein | http://paste.fedoraproject.org/50118/38568138/ |
| 09:22.59 | hos7ein | and |
| 09:23.12 | hos7ein | file /etc/asterisk/extensions.conf |
| 09:23.40 | hos7ein | http://paste.fedoraproject.org/50119/83038614/ |
| 09:23.59 | hos7ein | but i can not regestered clients ! |
| 09:24.06 | hos7ein | how to problems!? |
| 09:27.04 | hos7ein | * what is the problems? |
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| 10:07.39 | gavimobile | I own DID'S starting from 722215970 to 722215979. can I do [0-9] as follows to accept all calls from this number? exten => 72221597[0-9],1,Answer() |
| 10:08.47 | WIMPy | Almost. |
| 10:08.57 | WIMPy | If you add a _ in front, it will work. |
| 10:09.11 | WIMPy | Also X is an alias for [0-9] |
| 10:09.30 | gavimobile | WIMPy: _ apperently is my answer |
| 10:09.31 | gavimobile | thanks |
| 10:10.49 | gavimobile | WIMPy: worked perfectly |
| 10:12.42 | skrusty | morning |
| 10:13.04 | gavimobile | Patterns always start with an underscore (_). This tells Asterisk that we’re matching on a pattern, and not on an explicit extension name. |
| 10:13.31 | gavimobile | I for some reason thought they were for dialing out, not handling things coming in but that would be so off logic :-p |
| 10:13.35 | gavimobile | thanks again |
| 10:14.04 | WIMPy | Extensions are always incomming. |
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| 10:29.40 | gerritfromsa | hos7ein : start to remove the deny and permit statements |
| 10:30.46 | gerritfromsa | And this will NOT work : exten => 200,1,Dial(SIP/XXX) it must be : exten => 200,1,Dial(SIP/${EXTEN}) or simply exten => 200,1,Dial(SIP/200) |
| 10:31.13 | gerritfromsa | You can replace both lines with exten => _2XX,1,Dial(SIP/${EXTEN}) |
| 10:31.45 | gerritfromsa | This will allow you to dial 200-299 |
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| 10:55.04 | hos7ein | gerritfromsa: tnx, so, what is the : |
| 10:55.06 | hos7ein | Dial(SIP/${EXTEN}) |
| 10:55.07 | hos7ein | ? |
| 10:55.23 | hos7ein | <PROTECTED> |
| 10:55.25 | hos7ein | ? |
| 10:56.28 | gerritfromsa | The VARIABLE ${EXTEN} refers to what was matched in the 1st column (200 or 201) and dials what was matched in the 1st column |
| 10:57.15 | gerritfromsa | So if you has : exten => 200,1,Dial(SIP/${EXTEN}) it will dial sip client 200 when you dial 200 etc. |
| 11:02.26 | hos7ein | gerritfromsa: tnx |
| 11:02.38 | *** join/#asterisk atcmmi (~atcmmi@50.2.43.42) |
| 11:03.47 | magespawn | good day all |
| 11:05.49 | gerritfromsa | hos7ein : also remove nat=yes if these devices is on the same subnet |
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| 11:11.39 | ghost75 | when using chan_mobile is it possible to see the callerid ? it just displays name of mobile |
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| 11:15.26 | magespawn | i need to have two sip door phones call a certain extension rather than the main pbx |
| 11:16.51 | magespawn | i might add i am very new to working on asterisk |
| 11:18.12 | magespawn | i have calls going to and from the door phones, and to and from the extension, but would like the door phones to only call the extension |
| 11:18.24 | magespawn | any help or direction is appreciated |
| 11:19.20 | ghost75 | let the door dial the extension |
| 11:20.21 | magespawn | i did try that, it has that setting on the door phone itself, but still come through to the main switchboard |
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| 11:33.11 | magespawn | brb going to test something |
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| 12:17.28 | magespawn | right back |
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| 12:32.56 | magespawn | here is my exact problem with as much detail as i can think to add http://slexy.org/view/s2bm6ZKtP7 |
| 12:34.49 | magespawn | any help or direction appreciated |
| 12:35.51 | [TK]D-Fender | magespawn: Show us the actual debug for the call, not a vague description of it |
| 12:36.42 | magespawn | is it easier to get that from a terminal via ssh rather than putty? |
| 12:36.59 | [TK]D-Fender | Putty is best |
| 12:37.16 | [TK]D-Fender | and PuTTY *is* SSH |
| 12:37.54 | magespawn | yes i meant rather from a linux terminal or from PuTTY on windows? |
| 12:39.24 | [TK]D-Fender | PuTTY is best. Set a big buffer, right-click to copy all to buffer and paste away. No need to drag a selection |
| 12:45.45 | gerritfromsa | magespawn : does your door phones support autodial ? |
| 12:46.56 | gerritfromsa | If not you can put it in its own [context] and make it dial the 2 extensions regardless of what it dials |
| 12:47.03 | gerritfromsa | [doorphone] |
| 12:47.08 | [TK]D-Fender | gerritfromsa: He's got an error. Lets just wait for the actual debug. |
| 12:47.12 | magespawn | gerritfromsa, it has keys and button programming which can be set to dial an extension which you specify |
| 12:47.16 | magespawn | hold a sec |
| 12:47.32 | gerritfromsa | exten => _X.,1,Dial(SIP/1200$SIP/1201,,r) |
| 12:47.46 | Penguin | fail |
| 12:47.50 | [TK]D-Fender | yup |
| 12:48.15 | gerritfromsa | Door phones are so unreasonably priced ! |
| 12:48.36 | kaldemar | $ is a bad replacement for &. :) |
| 12:48.52 | gerritfromsa | Well spotted |
| 12:50.14 | Penguin | I can't believe this problem hasn't be resolved by now. But I am interested in why and how the problem exists. |
| 12:51.38 | magespawn | going to make the call |
| 12:51.56 | magespawn | Penguin I did have few other things come up, long story |
| 12:53.28 | kaldemar | there probably is no asterisk problem to resolve. |
| 12:54.46 | magespawn | probable not, my lack of knowledge is almost certainly the cause here |
| 12:55.48 | gerritfromsa | we all started that way |
| 12:56.46 | *** join/#asterisk nix8n82 (~AndChat27@24.143.11.81) |
| 12:57.32 | magespawn | http://slexy.org/view/s20Z63OWti hope i have done this right now |
| 12:59.00 | [TK]D-Fender | magespawn: there is no call in there |
| 12:59.54 | gerritfromsa | does anyone know how to disable ipv6 dns queries for bash |
| 13:01.32 | magespawn | [TK]D-Fender, that is what happens, no call from the door phone, did you see my earlier paste about the edits i have done to sip.conf and extensions.conf? http://slexy.org/view/s2bm6ZKtP7 |
| 13:02.20 | [TK]D-Fender | magespawn: "I get the notice in the Asterisk CLI call from 106 to 100 not allowed, because 100 not found in context." |
| 13:02.31 | [TK]D-Fender | magespawn: you said you had an error... here there is NO CALL AT ALL |
| 13:02.35 | [TK]D-Fender | magespawn: Apples & oranges |
| 13:02.59 | [TK]D-Fender | magespawn: A call previously reached your server... so where is your new attempt? |
| 13:03.26 | [TK]D-Fender | magespawn: Or you have possibly broken your doorphone's setup since then and now it can't talk at all... |
| 13:03.29 | magespawn | as it is currently i can call from 104 to 105 or but not from 105 or 106 to 104 |
| 13:03.53 | [TK]D-Fender | magespawn: Show us a failed call |
| 13:04.58 | magespawn | if i remove the context [intercom] from extensions.conf then they call the main switchboard which is 100 |
| 13:05.58 | gerritfromsa | they refer to the [default] context then ... |
| 13:06.11 | [TK]D-Fender | Do they? |
| 13:06.13 | [TK]D-Fender | I see nothing |
| 13:06.17 | magespawn | let me see what that says |
| 13:06.34 | magespawn | i think he means if i remove context [intercom] |
| 13:06.46 | [TK]D-Fender | Don't |
| 13:07.07 | [TK]D-Fender | You have configs in a documented state. USE them and show what actually happens |
| 13:07.10 | Greenlight | But there's no calls happening, what different is removing context's going to make |
| 13:07.59 | *** part/#asterisk mjordan (~mjordan@75.76.55.191) |
| 13:08.55 | StaRetji | I am using storesipcause=yes use_q850_reason=yes |
| 13:09.05 | StaRetji | sip show settings proves it, but I still don |
| 13:09.36 | StaRetji | see hangupcause, I must be dummest noob on earth |
| 13:10.49 | magespawn | [TK]D-Fender, sorry I am not even sure what that means |
| 13:11.12 | StaRetji | I thought exten => _X.,n,Hangup(${HANGUPCAUSE}) will be significant |
| 13:11.53 | Greenlight | You can provide a cause that way. |
| 13:12.00 | Penguin | Sometimes the hangupcasuse doesn't even matter. |
| 13:12.11 | *** join/#asterisk prasha (~prashant@123.236.189.20) |
| 13:12.13 | Penguin | s/hangupcasuse/hangupcause/ |
| 13:12.19 | [TK]D-Fender | magespawn: Place a call. Get actual debug. Doesn't matter what your configs say if calls aren't hitting your server at all. |
| 13:12.37 | [TK]D-Fender | magespawn: We need actual evidence. |
| 13:12.54 | magespawn | if i place a call from 104 to 105 will that help? |
| 13:13.02 | Greenlight | StaRetji: This is a snippet from a dialplan that outputs the hangup cause of a SIP channel. Verbose(1,#### Call via ${CDR(dstchannel)} to ${EXTEN} using ${server} failed (${DIALSTATUS}) - (${HANGUPCAUSE}) ${HANGUPCAUSE(${CDR(dstchannel)},tech)}. Trying any other unattempted entries in current priority ####); |
| 13:13.04 | [TK]D-Fender | magespawn: What should that show us? |
| 13:13.21 | Penguin | Is that where the problem is? If not, then how would it help? |
| 13:13.35 | [TK]D-Fender | magespawn: If I show you a picture of my cat, will that help my mechanic fix my transmission? |
| 13:13.39 | StaRetji | thx Greenlight, let me try swallow this lol |
| 13:13.50 | [TK]D-Fender | magespawn: Generate a failed call. |
| 13:14.04 | magespawn | [TK]D-Fender, lol, no it won't |
| 13:14.32 | magespawn | [TK]D-Fender, okay maybe i am not using the correct language to describe my problem |
| 13:14.45 | Greenlight | StaRetji: The output would be something like: Call via SIP/sip-wholesale-008361a4 to 0121XXXXXXX using sip-wholesale08.XXX.com completed (CONGESTION) - (19) SIP 480 Temporarily Unavailable |
| 13:14.50 | [TK]D-Fender | magespawn: You shouldn't be using any language at all. |
| 13:15.01 | [TK]D-Fender | magespawn: We need to see the ACTUAL DEBUG of your actual failed call. |
| 13:15.21 | [TK]D-Fender | magespawn: Stare at configs all day if you want, but you aren't going to prove your car works until you START THE ENGINE |
| 13:15.33 | [TK]D-Fender | magespawn: So show us that it WORKS by actually placing a call. |
| 13:15.47 | magespawn | okay hold on then |
| 13:15.57 | StaRetji | Greenlight: got it, now I need to understand how to reverse this in to extension, right? |
| 13:16.13 | Greenlight | StaRetji: What do you mean ? |
| 13:16.22 | StaRetji | or the one I showed above should produce that output? |
| 13:16.36 | StaRetji | yes, sorry, stupuid question, just realized that |
| 13:16.59 | Greenlight | Just put it after dial. |
| 13:17.39 | Greenlight | Since it's Dial that sets the hangupcause variable |
| 13:17.44 | StaRetji | I have no dial, I using a2billing, I've placed it at the end, is that wrong? |
| 13:17.55 | StaRetji | I'm using* |
| 13:18.08 | Greenlight | If you have no dial, then how are you making calls ? |
| 13:18.42 | Penguin | If there's no call, can there be a hangup? If there's no hangup, why should there be a hangup cause? |
| 13:18.55 | Greenlight | +1 |
| 13:19.01 | StaRetji | AGI(a2billing.php,1) |
| 13:19.18 | Greenlight | So do you Dial inside your AGI ? |
| 13:19.35 | StaRetji | well, I don't know mate, i just pass it to a2billing |
| 13:19.48 | Greenlight | I don't know either... |
| 13:19.49 | StaRetji | have no clue how to affect call after it is passed to it |
| 13:19.59 | Penguin | If YOU don't know what you're doing, how are we supposed to know? |
| 13:20.04 | Greenlight | Indeed... |
| 13:20.16 | gerritfromsa | StaRetji : Maybe wireshark would be better place to start |
| 13:20.18 | Greenlight | In these parts we call it "pissing into the wind" |
| 13:20.21 | StaRetji | well, I am sending to a2billing, then there is Hangup(${HANGUPCAUSE}) |
| 13:20.43 | [TK]D-Fender | And? |
| 13:20.56 | StaRetji | I am watching cli, to see once call is hangup |
| 13:21.00 | StaRetji | to get hangupcause |
| 13:21.12 | [TK]D-Fender | ... |
| 13:21.15 | StaRetji | and I am watching cdr too |
| 13:21.25 | StaRetji | but is only says FAILED |
| 13:21.32 | StaRetji | it* |
| 13:21.34 | [TK]D-Fender | does it? |
| 13:21.47 | [TK]D-Fender | I didn't see that.... |
| 13:21.50 | gerritfromsa | That won't get you the answer , cause YOU are issuing the HangUp() |
| 13:21.50 | Greenlight | You won't see the hangupcause in the CLI |
| 13:21.55 | [TK]D-Fender | Actuall... I still don't... |
| 13:22.12 | Greenlight | And yes, if YOU are hanging up, then YOU provide the cause... |
| 13:22.25 | magespawn | [TK]D-Fender, http://slexy.org/view/s2ZknfZLHU how is that? |
| 13:22.35 | gerritfromsa | Try exten => h,1,NoOp(${HANGUPCAUSE}) - executes this line on hangup |
| 13:22.41 | StaRetji | hm, but in wiki, ist say if I want to get hangup cause, that's the one |
| 13:22.58 | Greenlight | StaRetji: Link please |
| 13:23.07 | gerritfromsa | Not sure how you're gonna hack the PHP though ... |
| 13:23.10 | [TK]D-Fender | magespawn: that is not a complete call, and I see DAHDI in play now too.. |
| 13:23.47 | magespawn | [TK]D-Fender, yes there was an incoming call at the same time, do you need just one call happening |
| 13:23.49 | magespawn | ? |
| 13:23.50 | [TK]D-Fender | StaRetji: The WIKI isn't psychic about what other dialplan you have in play and you don't seem to be seriously looking |
| 13:23.57 | davlefouAMD | Bonjour, when i user file register with sql realtime user, i have no register, but it work with file user. Why and how solve the problem? |
| 13:24.11 | [TK]D-Fender | magespawn: I do not see a COMPLETE call. as in ALL of one from beginning to end. |
| 13:24.22 | [TK]D-Fender | magespawn: "core set verbose 10" <- while you're at it |
| 13:25.50 | *** join/#asterisk Draecos (~Draecos@106-69-55-54.dyn.iinet.net.au) |
| 13:26.18 | magespawn | [TK]D-Fender, i was following this for setting verbose and debug, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 13:26.21 | StaRetji | http://www.aa-asterisk.org.uk/index.php/Hangup_and_HANGUPCAUSE |
| 13:26.31 | *** join/#asterisk Draecos (~Draecos@106-69-55-54.dyn.iinet.net.au) |
| 13:26.39 | StaRetji | To read the cause code back you can use the ${HANGUPCAUSE} variable. If you receive a hangup cause from a channel and wish to pass it down the chain to another device you can do: exten => s,1,Hangup(${HANGUPCAUSE}) |
| 13:26.51 | *** join/#asterisk Draecos (~Draecos@106-69-55-54.dyn.iinet.net.au) |
| 13:26.56 | Greenlight | Thta's right. |
| 13:26.56 | [TK]D-Fender | magespawn: do as I have shown and provide a new call |
| 13:27.06 | [TK]D-Fender | StaRetji: Show us. |
| 13:27.23 | StaRetji | well, in my setup, I had exten => s,1,Hangup() now is exten => s,1,Hangup(${HANGUPCAUSE}) |
| 13:27.36 | StaRetji | [TK]D-Fender: show what? |
| 13:27.42 | [TK]D-Fender | StaRetji: ... the call. |
| 13:27.43 | Greenlight | SO you are trying to pass the hangupcause to the remote side ? |
| 13:27.56 | StaRetji | no, I am trying to log it |
| 13:27.59 | StaRetji | to cdr |
| 13:28.03 | Greenlight | So, why are you doing that. |
| 13:28.10 | Greenlight | Read the sentance you just pasted. |
| 13:28.11 | [TK]D-Fender | "Do or do not, there is no try" - Yoda |
| 13:28.46 | StaRetji | so, if I want to log hangupcause in cdr, I don't actually need any of the above? |
| 13:29.45 | [TK]D-Fender | StaRetji: Who says that variable ever gets saved there? |
| 13:30.42 | Greenlight | StaRetji: First, just do a NoOp and spit it out to the CLI. Then once your happy, save that to the CDR. |
| 13:31.29 | StaRetji | hmm, okay, will try that now, thx for patient with me, I am sorry, I don't do this cause I am lazy or something, I just don't get it pretty well |
| 13:32.28 | magespawn | [TK]D-Fender, http://slexy.org/view/s20VwVs6ed |
| 13:32.45 | Greenlight | You problem is that you don't know what your dialplan is doing, since your passing control to a 3rd party AGI. |
| 13:33.31 | magespawn | Greenlight, me? |
| 13:33.42 | Greenlight | No, StaRetji. Sorry I should have been specific. |
| 13:33.43 | [TK]D-Fender | magespawn: No, not you |
| 13:34.01 | [TK]D-Fender | magespawn: Why is there no SIP debug in this one? |
| 13:34.27 | StaRetji | NoOp(SIP return code : ${HASH(SIP_CAUSE,${CDR(dstchannel)})}) |
| 13:34.34 | StaRetji | I guess this is it? |
| 13:34.43 | [TK]D-Fender | magespawn: And in this debug I see 2 calls being placed, both being answered. What is WRONG in this dump? |
| 13:34.57 | magespawn | [TK]D-Fender, sorry did not set it, only set verbose to 10 |
| 13:35.13 | magespawn | i need the second call to go to extension 104 |
| 13:35.14 | [TK]D-Fender | magespawn: Still, what am I supposed to be seeing in there? I see 2 calls that seemed to process fine. |
| 13:35.30 | Greenlight | StaRetji: Start with just ${HANGUPCAUSE} |
| 13:35.59 | Greenlight | Then if you want the more detailed channel tech specific cause, use ${HANGUPCAUSE(${CDR(dstchannel)},tech)} |
| 13:35.59 | [TK]D-Fender | magespawn: What second call? I see SIP/100 dialing 150, which rings SIP/150 |
| 13:36.26 | [TK]D-Fender | magespawn: Then I see SIP/106 dialing 100, which rings SIP/100 |
| 13:36.39 | [TK]D-Fender | magespawn: BOTH seemed to actually call and actually get answered. |
| 13:36.52 | [TK]D-Fender | Mage nothing seems out of the ordinary there. So what's the problem? |
| 13:37.35 | magespawn | [TK]D-Fender, yes they do but i need the 106 call to go to 104 not 100 |
| 13:37.50 | Penguin | Dial 104, then. |
| 13:38.00 | Greenlight | I admire your patience. |
| 13:38.09 | [TK]D-Fender | magespawn: 106 dialed 100. that is YOUR doing. |
| 13:38.29 | [TK]D-Fender | magespawn: You .... on that device ... made it dial 100 |
| 13:38.39 | [TK]D-Fender | magespawn: that has nothing to do with ASTERISK |
| 13:38.40 | Penguin | It clearly says SIP/106 dialed extension 100, which rings SIP/100. |
| 13:39.05 | [TK]D-Fender | magespawn: if that is your door-phone and it is an "automatically dialed number", then you gave it the WRONG NUMBER |
| 13:39.23 | Penguin | That's what I told him last week. |
| 13:39.32 | magespawn | Penguin, [TK]D-Fender, the door phone has a single button that you can program to call an extension which i set to 104 but it still come out at 100 |
| 13:39.35 | Greenlight | remembers |
| 13:39.49 | [TK]D-Fender | magespawn: you programmed that button wrong. We see the call right here. |
| 13:40.12 | [TK]D-Fender | magespawn: Asterisk is not misinteerpreting the door-phone. You simply configured it wrong. |
| 13:40.17 | [TK]D-Fender | magespawn: Go fix your door-phone |
| 13:40.22 | magespawn | not according to the web interface of the door phone |
| 13:40.24 | [TK]D-Fender | magespawn: It is wrong. |
| 13:40.33 | [TK]D-Fender | magespawn: According to asterisk debug, it is. |
| 13:40.36 | [TK]D-Fender | magespawn: Look again. |
| 13:40.49 | magespawn | [TK]D-Fender, will do |
| 13:40.50 | [TK]D-Fender | magespawn: If you want a sanity check, show US while you're at it |
| 13:41.33 | magespawn | [TK]D-Fender, how to do that? |
| 13:41.54 | [TK]D-Fender | magespawn: through the magic of SCREEN-SHOTS |
| 13:42.09 | [TK]D-Fender | magespawn: they come in pretty pixels of all colours |
| 13:42.27 | gerritfromsa | magespawn : Maybe a typo in the dialplan ? ex : exten => 104,1,Dial(SIP/100) ? |
| 13:42.31 | [TK]D-Fender | imgur , tinypic , whatever. Your own host. Whatever tool you want. |
| 13:42.38 | magespawn | [TK]D-Fender, lol okay |
| 13:42.40 | [TK]D-Fender | gerritfromsa: NO. |
| 13:42.46 | Greenlight | It dialled 100 |
| 13:42.53 | Greenlight | Said so in his paste |
| 13:42.55 | Penguin | gerritfromsa: No. The pastebin clearly shows 100. |
| 13:42.56 | [TK]D-Fender | gerritfromsa: He's dialing the wrong number from the door-phone. |
| 13:43.00 | gerritfromsa | Gotit |
| 13:43.49 | zpotoloom | magespawn, have you tried rebooting the door-phone after changing the settings |
| 13:44.04 | magespawn | zpotoloom, yes |
| 13:44.11 | Greenlight | wonders if exten => 100/106,1,Dial(SIP/104) would be an easy fix |
| 13:44.14 | magespawn | let me get screen shots |
| 13:44.18 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 13:44.19 | gerritfromsa | why is it a windows phone :) |
| 13:44.25 | Penguin | gerritfromsa: No. |
| 13:44.28 | [TK]D-Fender | Greenlight: Don't go there |
| 13:44.43 | [TK]D-Fender | Greenlight: Don't enable "stupid" :p |
| 13:44.47 | Greenlight | I just don't know how much longer your patience can hold out! |
| 13:45.06 | Penguin | Just until it is fixed. |
| 13:47.17 | gerritfromsa | <PROTECTED> |
| 13:47.25 | Penguin | Don't do it. |
| 13:47.53 | magespawn | what is the preferred service for picture uploading? |
| 13:48.08 | Penguin | Fix the PROBLEM rather than convolute the dial plan. |
| 13:48.16 | Penguin | magespawn: imagebin.org is fine. |
| 13:51.16 | Penguin | There's also picpaste.com. |
| 13:52.36 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 13:53.36 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 13:57.54 | magespawn | here http://imagebin.org/275138 and http://imagebin.org/275139 |
| 13:58.23 | magespawn | [TK]D-Fender, ^^ |
| 13:59.00 | [TK]D-Fender | magespawn: Show us a new call with SIP debug enabled |
| 13:59.06 | magespawn | okay |
| 13:59.31 | magespawn | so leave verbose to 10 and set sip debug on? |
| 14:00.31 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:00.31 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:01.01 | magespawn | brb |
| 14:01.08 | Greenlight | That appeared to be the config for "key 0", is that definetly what's being pressed ? |
| 14:01.28 | Penguin | I was wondering how many keys it has. |
| 14:01.42 | Greenlight | http://www.doorphones.co.za/surix-ip-multi-button-door-phone/ ? |
| 14:01.49 | Greenlight | That one has a few |
| 14:01.57 | magespawn | Greenlight, it is the only button, but there is no indication if they start the numbering at 0 or 1 |
| 14:02.19 | magespawn | i have this one http://www.miro.co.za/detail.aspx?pid=2066&p=VoIP&sp=IP%20Phones&spp=Door%20Phones&spid=187&sspid=669 |
| 14:02.25 | Greenlight | magespawn: Doesn't it come with an instruction book ? |
| 14:02.49 | Penguin | If you have a program for key 0 and for key 1, program them BOTH to call extension 104. |
| 14:02.53 | magespawn | Greenlight yes but not very clear, translated from spanish |
| 14:03.02 | magespawn | Penguin, good idea |
| 14:03.03 | Greenlight | Do what Penguin suggests |
| 14:03.08 | magespawn | brb |
| 14:03.37 | Penguin | It could be a generic firmware image and your single-key phone uses key 1 instead of key 0. |
| 14:03.44 | Greenlight | Mhm |
| 14:04.02 | Greenlight | And 100 sounds very much like a "default" that it might call if not configured |
| 14:04.13 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-voqyjaqpfhgrzaqc) |
| 14:04.13 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:04.37 | boom^time | Greenlight, did you get a chance to test that outbound fax detection at all? |
| 14:04.40 | magespawn | okay now button 0 and 1 set to call extension 104 brb going to make a call |
| 14:04.54 | Greenlight | boom^time: Nope, but I will, don't worry! :) |
| 14:05.03 | boom^time | Greenlight, thanks! |
| 14:05.09 | [TK]D-Fender | magespawn: Oh so there was something that WASN'T pointed to where you wanted? |
| 14:05.27 | Greenlight | I had planned to have a look this morning, but been dragged into confernece calls, and now got a crazy workload |
| 14:05.45 | boom^time | Yeah I understand. |
| 14:05.47 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 14:06.04 | Greenlight | I've a few of my own projects that it would be quite handy for |
| 14:06.30 | Greenlight | Plus it's annoyed me that it wasn't working when it seemed like it should :) |
| 14:07.04 | boom^time | Right, it seems as though it should no problem. |
| 14:07.13 | Greenlight | Quite |
| 14:07.39 | Greenlight | I'm sure things will reveal themsevles with some extra debugging in there ;) |
| 14:07.48 | Penguin | I have a feeling the direction of the call matters. |
| 14:07.52 | *** join/#asterisk JamKo (~JamKo@unaffiliated/jamko) |
| 14:08.11 | Greenlight | Penguin: Yea, I suspect that might come into play, be it intentional or not. |
| 14:08.30 | Greenlight | The way the docs I was reading were worded it's supposed to work in both directions. |
| 14:09.15 | magespawn | [TK]D-Fender, Greenlight, Penguin here is the latest http://slexy.org/view/s20OHja8bv |
| 14:09.59 | Greenlight | Your door phone is 106, no ? |
| 14:10.10 | magespawn | [TK]D-Fender, there might still be, i did not write the dial plan |
| 14:10.13 | Penguin | There's no call in that paste. |
| 14:10.15 | magespawn | Greenlight, yes |
| 14:10.22 | Greenlight | Then, um, it's not there.. |
| 14:10.23 | [TK]D-Fender | magespawn: Dialplan? Huh? |
| 14:10.29 | [TK]D-Fender | magespawn: I'm talking about the SURIX |
| 14:10.35 | [TK]D-Fender | magespawn: this has nothing to do with Asteriask |
| 14:10.41 | [TK]D-Fender | magespawn: Stop changing the target. |
| 14:11.02 | [TK]D-Fender | magespawn: Before it had one button. Then you say TWO. We never see screenshots proving any of this srecond part you then brought up |
| 14:11.10 | [TK]D-Fender | magespawn: and this las debug was not a complete call |
| 14:11.37 | [TK]D-Fender | magespawn: User-Agent: SURiX iP Access V 01.98 From: "104" <sip:104@10.0.0.254>;tag=as7188ae16 To: <sip:105@10.0.0.50:5060> |
| 14:11.50 | [TK]D-Fender | magespawn: We do seem to see a comm that impied that it at least tried to call 105..... |
| 14:11.56 | [TK]D-Fender | (supposedly) |
| 14:12.02 | Greenlight | Oh, the doorphone is 103 |
| 14:12.03 | Greenlight | 104 |
| 14:12.24 | Penguin | 105, 106? |
| 14:12.29 | magespawn | no the doorphones are 105 and 106 |
| 14:12.37 | Greenlight | There are *two* ? |
| 14:12.41 | magespawn | the phone i want them to call is 104 |
| 14:12.51 | [TK]D-Fender | [09:37]magespawn[TK]D-Fender, yes they do but i need the 106 call to go to 104 not 100 |
| 14:13.05 | magespawn | Greenlight, yes one for entry and one for exit |
| 14:13.14 | Greenlight | Okay. |
| 14:13.17 | [TK]D-Fender | magespawn: show a complete call. |
| 14:13.20 | Penguin | magespawn: By the way, those are TERRIBLE names for phone sets. |
| 14:13.47 | Greenlight | But in that latest paste, 104 seems to *be* a door phone |
| 14:13.58 | magespawn | Penguin, agreed, i am just working with the way the original plan was set |
| 14:14.03 | Greenlight | Unless i'm being stupid and misreading it |
| 14:14.44 | Penguin | It would be easier to read if they were named appropriately. |
| 14:15.03 | Penguin | If you saw SIP/FrontDoor, you'd know which set was doing the calling. |
| 14:16.41 | Greenlight | We seem to catch the end of a call, as it's being CANCELed. BUt that call seems to be *from* 104 *to* 105 |
| 14:16.46 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 14:20.39 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 14:23.31 | magespawn | okay have renamed them, i will go make another call |
| 14:23.42 | [TK]D-Fender | <--- SIP read from UDP:10.0.0.50:5060 ---> <--- call was from this SURIX |
| 14:23.52 | Penguin | Let's see if we get less ambiguity now. |
| 14:23.53 | [TK]D-Fender | the images were from 10.0.0.51 |
| 14:23.56 | [TK]D-Fender | NOT the same device |
| 14:24.03 | [TK]D-Fender | Someone's head is not on straight at all |
| 14:24.15 | [TK]D-Fender | And we're talking about what appears to be a FOUR device PBX |
| 14:24.27 | magespawn | [TK]D-Fender, that maybe true |
| 14:25.42 | Penguin | In any case, if both dork phones were configured the same way, pressing the key on either of them *should* result in a call to the same extension. |
| 14:26.53 | [TK]D-Fender | 2 devices with apparently now 2 buttons to configure. You'd like to think we'd be running out of wrong combinations by now. |
| 14:28.52 | magespawn | [TK]D-Fender, there has always been two door phones, one physical button on each but with a generic web interface that has the programming for the multibutton door phone |
| 14:29.51 | [TK]D-Fender | magespawn: Anything new for us to see? |
| 14:30.08 | magespawn | http://slexy.org/view/s2NZkKVSmU |
| 14:31.49 | [TK]D-Fender | magespawn: Nothing about your door-phones in there that I can see |
| 14:32.19 | [TK]D-Fender | magespawn: Also appears you didn't heed my previous advice and increase the SCROLL-BACK BUFFER |
| 14:32.31 | [TK]D-Fender | magespawn: Right-click, go into your options and increase the number of lines |
| 14:33.16 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-oiilafqnbyrxhuoh) |
| 14:33.17 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:33.25 | *** join/#asterisk zigg (~matt@unaffiliated/zigg) |
| 14:34.16 | magespawn | right trying that again |
| 14:40.07 | magespawn | here we go http://slexy.org/view/s2XDRsoCFj |
| 14:41.22 | Penguin | You have a phone named 104 and a device in asterisk named Reception. |
| 14:43.22 | magespawn | Penguin, yes they are the same thing |
| 14:43.32 | magespawn | have i made a mistake there? |
| 14:43.36 | Penguin | Asterisk says they are not the same thing. |
| 14:43.46 | Penguin | The PHONE thinks it is 104. |
| 14:43.58 | Penguin | Asterisk says you have a device configuration for Reception. |
| 14:44.15 | *** join/#asterisk danilom (~danilom@190.38.26.230) |
| 14:44.15 | magespawn | okay so i need to change the phone to match asterisk? |
| 14:44.24 | danilom | hi all |
| 14:44.45 | Penguin | At least now I see exitdoor is calling extension 100. |
| 14:44.58 | Penguin | And extension 100 is dialing a phone by the name of SIP/100. |
| 14:45.12 | [TK]D-Fender | [10:12][TK]D-Fender[09:37]magespawn[TK]D-Fender, yes they do but i need the 106 call to go to 104 not 100 |
| 14:45.17 | [TK]D-Fender | Penguin: It was always going there |
| 14:45.22 | [TK]D-Fender | Penguin: and that is what is BAD |
| 14:45.35 | Penguin | It's bad, but the last one didn't show any call at all. |
| 14:46.20 | [TK]D-Fender | <--- SIP read from UDP:10.0.0.51:5060 ---> INVITE sip:100@10.0.0.254 ---- From: <sip:SIP/exitdoor@10.0.0.51>;tag=3124143274 ----SIP/2.0 User-Agent: SURiX iP Access V 01.98 |
| 14:47.44 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 14:48.13 | danilom | im having connection refused form asterisk, i check the manager.conf, also test doing telnet localhost 5038, im using asterisk 11 in debian, any idea where to look at? |
| 14:48.13 | Nugget | telnet is eeeeeeevil! |
| 14:48.24 | Penguin | If programming _all_ the keys to call extension 104 still results in calls to 100, the only other thing I can think of is accept that the phones are broken and deal with it. |
| 14:48.47 | Greenlight | Or search for updated/fixed firmware |
| 14:48.56 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.244.113) |
| 14:48.57 | gerritfromsa | port 23 could be firewalled |
| 14:49.06 | Greenlight | My earlier suggesting is looking more and more promising though... |
| 14:49.11 | Penguin | gerritfromsa: Manager doesn't use port 23. |
| 14:49.22 | Greenlight | AMI is on 5038 by default |
| 14:49.30 | gerritfromsa | Sorry - port 5038 could be firewalled |
| 14:49.41 | Penguin | greenlight: It's starting to look better. |
| 14:49.44 | danilom | i check iptables |
| 14:49.55 | Greenlight | danilom: Have you restarted asterisk since changing manager.conf ? |
| 14:50.05 | Penguin | Why would you? |
| 14:50.09 | Penguin | Just load/reload the manager module. |
| 14:50.20 | gerritfromsa | also check deny and permit in manager.conf |
| 14:50.23 | danilom | Greenlight, well i restart all the machine |
| 14:50.24 | Greenlight | Restarting is less prone to "user error" |
| 14:50.37 | Penguin | It's also totally unnecessary. |
| 14:50.43 | danilom | Penguin, yes |
| 14:50.44 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:50.44 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:51.04 | Greenlight | Indeed, but it's an easy way to make sure this ins't a config-not-loaded issue |
| 14:51.05 | Penguin | That's like restarting your computer to close your browser. |
| 14:51.15 | Penguin | JUST CLOSE THE BROWSER! |
| 14:51.16 | Ice_Strike | Going to upgrade 1.4.36 to 1.8 ;) |
| 14:51.22 | Greenlight | WHy |
| 14:51.28 | danilom | Penguin, is a VM doesnt matter |
| 14:51.29 | Greenlight | 11 is newer |
| 14:51.36 | Penguin | false |
| 14:51.40 | danilom | ok, |
| 14:52.02 | [TK]D-Fender | Ice_Strike: Why moving to an older LTS? |
| 14:52.12 | danilom | sorry but i dont know where to llok, i have a cli open but not show nothing |
| 14:52.27 | [TK]D-Fender | Ice_Strike: You are losing time in the support window and less functionality |
| 14:52.42 | Penguin | danilom: Did you enable manager in manager.conf and load or reload the module? |
| 14:53.05 | [TK]D-Fender | Are you looking in TCPDUMP to prove packets are even arriving? |
| 14:53.12 | danilom | Penguin, i have enable=yes and the user added in the manager.conf too |
| 14:53.25 | Penguin | danilom: Did you check to see that manager is listening? |
| 14:53.41 | Penguin | lsof -i :5038 |
| 14:53.43 | danilom | sorry, how can i test if manager is listening? |
| 14:54.00 | Greenlight | Did you allow the IP in manager.conf ? |
| 14:54.14 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 14:54.16 | danilom | Greenlight, yes im testing from localhost |
| 14:54.29 | Greenlight | So, you've permit=127.0.0.1 ? |
| 14:54.32 | Greenlight | *after* the deny ? |
| 14:54.42 | Ice_Strike | I could try 11.2-cert2 but Is there dial plan chnages from 1.4 ? |
| 14:54.48 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 14:54.53 | Ice_Strike | Not features, I meant changes existing dial plan |
| 14:55.04 | [TK]D-Fender | Ice_Strike: 1.6 has changes from 1.4.... |
| 14:55.06 | Penguin | If manager is not listening, nothing else you do will help. Start with the beginning and work your way out. |
| 14:55.17 | [TK]D-Fender | Ice_Strike: You're already about the make changes... make them the RIGHT ones |
| 14:55.33 | Greenlight | Indeed, I can see up reason to "upgrade" to 1.8 now. |
| 14:55.37 | Greenlight | Go with 11 |
| 14:55.40 | danilom | Greenlight, yes is after the deny line |
| 14:56.14 | danilom | Penguin, lsof -i :5038 dont show anything, how can i check if module is running? |
| 14:56.21 | Ice_Strike | Alright, ;) I simply uninstall 1.4 and install 11? |
| 14:56.28 | Penguin | If it shows nothing, manager is not listening. |
| 14:56.30 | Ice_Strike | I will keep the backup of /eta/asterisk/ |
| 14:56.36 | Greenlight | First, yes, backup. |
| 14:56.41 | Ice_Strike | Hopefully it will work with 11 |
| 14:56.44 | Ice_Strike | using same one. |
| 14:56.48 | Greenlight | Not sure what you mean by "uninstall". |
| 14:56.54 | danilom | Penguin, ok im going to read a bit more to learn how |
| 14:56.58 | Greenlight | Read the change notes for the releases |
| 14:57.09 | Greenlight | You *will* need to make changes from 1.4 |
| 14:57.12 | danilom | how to start the module |
| 14:57.36 | magespawn | Penguin, [TK]D-Fender any ideas? should i just shoot myself now and be done with it? |
| 14:57.58 | Ice_Strike | By uninstall, I mean remove 1.4 asterisk. |
| 14:58.02 | Greenlight | magespawn: Contact the company who make them |
| 14:58.27 | Greenlight | Or Google for a solution or firmware upgrade. |
| 14:58.46 | magespawn | Greenlight so you think the problem lies with the doorphones? |
| 14:58.53 | Penguin | danilom: Did you try "manager reload" on the ast CLI? |
| 14:58.54 | Greenlight | I don't think. I know. |
| 14:59.21 | magespawn | capital advice then i will do that. i will also let you all know how that goes |
| 14:59.22 | gerritfromsa | magespawn : or just change the dialplan ... |
| 14:59.24 | danilom | Penguin, im going to try sorry, im learning. |
| 14:59.26 | [TK]D-Fender | magespawn: we can see the device dialing something you don't want it to be dialing. we've seen all of 2 screenshots from it. That device is responsible for what it is dialing, not Asterisk. Fix the device. |
| 15:00.04 | [TK]D-Fender | gerritfromsa: Stop trying to advocate bad hacks. He isn't going to learn. |
| 15:00.14 | Penguin | magespawn: If the software interface has multiple keys for a single-key phone, did you program ALL of the keys to dial 104? |
| 15:00.20 | magespawn | [TK]D-Fender, i will try that, gives me a starting point at least |
| 15:00.25 | magespawn | no Penguin |
| 15:00.26 | [TK]D-Fender | gerritfromsa: And you are distracting from those who are trying to set him on the proper course |
| 15:00.39 | [TK]D-Fender | mageNO?! |
| 15:00.43 | Penguin | magespawn: Program every single key in the interface to dial 104. |
| 15:00.50 | [TK]D-Fender | magespawn: why the hell not?\ |
| 15:00.53 | Greenlight | ... |
| 15:00.54 | Penguin | magespawn: This is what we know as trial and error. |
| 15:00.55 | gerritfromsa | if the guys in a fix and its not like he didnt try , sometimes a fix is a fix |
| 15:01.09 | [TK]D-Fender | magespawn: You're wondering why it's dialing something you didn't want it to and left OTHER stuff in there? |
| 15:01.10 | Greenlight | Did't we decided to do that like an hour ago... |
| 15:01.56 | Penguin | If I ran into this, I would try key 0. If that didn't work, I would try key 1. If that didn't work, I would have tried key 2. If that didn't work, I would have tried key 3. |
| 15:01.57 | magespawn | okay [TK]D-Fender i will do that now, Greenlight i thought to only do key 0 and 1 |
| 15:02.13 | Greenlight | How many keys are there ? |
| 15:02.14 | magespawn | Penguin, lesson learned |
| 15:02.22 | magespawn | let me check |
| 15:02.29 | Greenlight | You TOLD US there were only 0 & 1 that could be set. |
| 15:02.31 | danilom | Penguin, i left the manager.conf like default but in CLI making manager reload show me |
| 15:02.32 | danilom | Unable to open AMI configuration manager.conf, or configuration is invalid. Asterisk management interface (AMI) disabled. |
| 15:02.55 | Greenlight | danilom: You have a typo in manager.conf most likely. Pastepin it. |
| 15:03.04 | danilom | ok |
| 15:03.10 | Ice_Strike | does "show agents" command work in asterisk 11? |
| 15:03.10 | Greenlight | Or a permissions issue, perhaps. |
| 15:03.13 | Penguin | If you program every single key to 104 and it STILL doesn't dial 104, then I would start to think about different firmware or even a possible work around. |
| 15:03.34 | Penguin | ice_strike: Don't expect show anything to work. |
| 15:03.50 | [TK]D-Fender | I don't even trust that he's updating the same device we've been talking about at this point. |
| 15:03.54 | Penguin | ice_strike: It's typically going to be: <component> show |
| 15:04.03 | magespawn | i think i have found it now, there is one called button 0ne and all the others are called key, completly missed that brb |
| 15:04.04 | [TK]D-Fender | No consistency, no follow-through, and no new evidence |
| 15:04.15 | Greenlight | My god. |
| 15:04.19 | Penguin | hahaha |
| 15:04.23 | danilom | i have pasted mymanager.conf here http://pastebin.com/U4mDbtwM |
| 15:04.24 | file | looks around |
| 15:04.43 | gerritfromsa | "possible work around" is what I was suggesting |
| 15:05.15 | Penguin | It's a last resort. Always. |
| 15:05.21 | Penguin | Fix the problem, don't work around it. |
| 15:05.25 | Greenlight | danjenkins: That looks okay. What are the permissions ? |
| 15:05.35 | Penguin | If you absolutely cannot fix it, THEN go around it. |
| 15:06.07 | danjenkins | think you mean danilom Greenlight ;) |
| 15:06.20 | Greenlight | Yes, sorry, auto-complete fail. |
| 15:06.29 | danilom | Greenlight, of the manager.conf file? -rw-rw-r-- 1 asterisk asterisk 1204 oct 29 10:33 manager.conf |
| 15:06.50 | file | replaces danjenkins with a cardboard cutout danjenkins |
| 15:07.00 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 15:07.08 | Penguin | That should be okay, but I would rather see world not be able to read a file with a password in it. |
| 15:07.12 | danjenkins | :( file |
| 15:07.46 | Greenlight | danilom: Did you edit this in Windows ? |
| 15:08.23 | gerritfromsa | danilom : youre trying to telnet in from the same server correct? |
| 15:08.25 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:08.27 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:09.13 | Greenlight | Nothing seems obviously wrong there. It's a longshot, but perhaps a funky character in there somewhere |
| 15:09.56 | [TK]D-Fender | remove all the extra white-space while you're at it |
| 15:10.00 | [TK]D-Fender | and the commented out junk |
| 15:10.23 | [TK]D-Fender | and it'd be nice to see what's actually loading... |
| 15:10.28 | [TK]D-Fender | "core show settings" |
| 15:10.32 | danilom | Greenlight, no im in debian |
| 15:10.32 | [TK]D-Fender | "module show" |
| 15:10.45 | [TK]D-Fender | And I don't trust PATHS here in the slightest either... |
| 15:10.56 | danilom | gerritfromsa, i try telnet in the same pc, |
| 15:11.56 | danilom | ok im goin to see |
| 15:12.21 | Greenlight | Either the error is misleading, or there's something wrong with that file which we can't see |
| 15:13.05 | Greenlight | if you've only edited in linux then I can't see funky characers sneaking their way in, but perhaps as [TK]D-Fender suggests whitespace is causing issues. |
| 15:13.43 | Greenlight | or option 3, you're showing us the wrong file :) |
| 15:13.46 | [TK]D-Fender | I don't see actual folders with actual files, nor what *'s running environment is. |
| 15:13.54 | [TK]D-Fender | trust = 0 |
| 15:14.09 | [TK]D-Fender | And that's only because it is an UNSIGNED INT |
| 15:14.18 | Greenlight | :) |
| 15:14.46 | danilom | ok, where can i see, if asterisk is looking in a diff path? |
| 15:15.03 | danilom | so im checking /etc/asterisk |
| 15:15.28 | Penguin | I don't see any modules in my own asterisk related to ami. And I have ami enabled and working. |
| 15:15.30 | danilom | i edited all the files with nano, because dont have X installed |
| 15:15.40 | Penguin | No ami, no manager. |
| 15:16.07 | Penguin | danilom: You don't need X to edit text files. So good move there. |
| 15:16.25 | Penguin | What nano wouldn't have been MY choice, it should work well enough. |
| 15:16.31 | Penguin | s/What/While/ |
| 15:16.56 | [TK]D-Fender | [11:14]danilomok, where can i see, if asterisk is looking in a diff path? [11:10][TK]D-Fender"core show settings" [11:10][TK]D-Fender"module show" |
| 15:17.13 | magespawn | Penguin, [TK]D-Fender , Greenlight , gerritfromsa it was my stupidity or assuming that I was programming the right interface, THANK YOU very much for all your time and effort |
| 15:17.38 | Penguin | magespawn: Now you found button programming? |
| 15:17.44 | Greenlight | laughs |
| 15:18.24 | Greenlight | Glad you got it working in the end though... |
| 15:18.25 | danilom | module show http://pastebin.com/av1B6qKA |
| 15:18.42 | magespawn | yup was programming the key which this does not have it only has a single physical button Penguin |
| 15:18.45 | Greenlight | and "core show settings" |
| 15:19.17 | magespawn | if you guys are ever in my part of the world i owe you about 100 beers each i reckon |
| 15:20.07 | Penguin | What part is that? |
| 15:20.12 | danilom | http://pastebin.com/P0CBynJg --> config directory /etc/asterisk |
| 15:20.17 | magespawn | South Africa, Hluhluwe. |
| 15:21.13 | [TK]D-Fender | danilom: "ls -la /etc/asterisk" |
| 15:22.23 | magespawn | [TK]D-Fender, thanks for the help |
| 15:22.57 | danilom | http://pastebin.com/05JgnvZd |
| 15:23.48 | [TK]D-Fender | magespawn: When we beat it into you that your device is choosing to do something... drill the shit out of EVERY config screen. |
| 15:24.06 | magespawn | [TK]D-Fender, lesson learned |
| 15:25.10 | [TK]D-Fender | danilom: You have "root" files in there. fix those, then delete the "bak" files or move them WAY out, and pastebin your new manager.conf and an attempt to restart * |
| 15:25.43 | danilom | ok thanks |
| 15:27.49 | Greenlight | Perhaps an "asterisk -cvvvvv" might be best? |
| 15:28.57 | Penguin | I usually chown asterisk:asterisk /etc/asterisk/*.conf && chmod 640 /etc/asterisk/*.conf |
| 15:29.43 | *** join/#asterisk Draecos (~Draecos@106-69-55-54.dyn.iinet.net.au) |
| 15:29.47 | magespawn | bye all, see you again on another day |
| 15:30.08 | *** join/#asterisk navaismo (~navaismo@189.191.200.240) |
| 15:30.44 | *** part/#asterisk magespawn (~Eames@105-236-71-217.access.mtnbusiness.co.za) |
| 15:35.24 | davlefouAMD | hi, i have that problem http://pastebin.com/vGHjiLSz , how can i solved it? |
| 15:36.11 | [TK]D-Fender | davlefouAMD: compile with the same version as the rest of your * was built with |
| 15:38.58 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 15:40.05 | davlefouAMD | i have that version : Asterisk 1.8.8.0~rc4-1digium0+1~lucid built by pbuilder @ nighthawk on a x86_64 running Linux on 2011-11-22 15:53:21 UTC |
| 15:40.26 | davlefouAMD | and get src of : /asterisk-1.8.8.0~rc4 |
| 15:40.38 | davlefouAMD | it seems me the same? |
| 15:40.45 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 15:40.49 | [TK]D-Fender | [Feb 15 22:09:26] WARNING[17463] loader.c: Module 'codec_ilbc.so' was not compiled with the same compile-time options as this version of Asterisk. |
| 15:40.56 | [TK]D-Fender | THIS module was not compiled with the same options |
| 15:41.44 | Penguin | Isn't ilbc provided by a different package? |
| 15:42.05 | davlefouAMD | http://pastebin.com/QU3sNa8K |
| 15:42.17 | Penguin | When I make a package, I have to include sources to ilbc from a different location. |
| 15:43.08 | davlefouAMD | yes, i get ilbc with the that shell : get_ilbc_source.sh |
| 15:43.16 | Penguin | Do you need ilbc? |
| 15:43.32 | Penguin | I think most people do not use it. |
| 15:43.41 | davlefouAMD | yes, i try to use it for my project. |
| 15:43.45 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 15:44.08 | danfromuk | Hi, we've been asked if we can support the speex codec. Are there any issues with using it? Do I need to do anything apart from allowing it? |
| 15:44.25 | davlefouAMD | it seems better than g729 for bad internet. |
| 15:45.10 | davlefouAMD | http://www.voip-info.org/wiki/view/iLBC |
| 15:48.12 | navaismo | whisper Opus |
| 15:53.05 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:53.06 | davlefouAMD | Under ubuntu 12.04, ilbc was in, so it is possible to compil it! |
| 15:53.21 | [TK]D-Fender | davlefouAMD: You did it wrong. |
| 15:53.22 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 15:53.44 | Ice_Strike | I've just upgraded 1.4 to 11 |
| 15:53.53 | Ice_Strike | Should I use new asterisk config |
| 15:54.02 | Ice_Strike | or use existing one from 1.4? |
| 15:54.07 | [TK]D-Fender | Ice_Strike: What new "asterisk config"? |
| 15:54.10 | Penguin | davlefouamd: If you have installed asterisk from a package, see if ilbc also is in a package. |
| 15:54.15 | LooserOuting | danfromuk: on speex you can set quality and bandwith. so maybe speex can be better then g729 but needs definetly more bandwith. The link is about iLBC wich is not speex. iLBC mode 20 is nice but asterisk only supports mode 30. AFAIK all 3 codecs have PLC |
| 15:54.26 | Ice_Strike | [TK]D-Fender /etc/asterisk/* |
| 15:54.41 | [TK]D-Fender | Ice_Strike: Try beng a little more vage and maybe remove every second preposition or cunjuction.... |
| 15:54.51 | danfromuk | Ice_Strike: there are loads of changes. Look at CHANGES for more info |
| 15:55.05 | [TK]D-Fender | Ice_Strike: did you read the upgrade.txt between versions to understand what the differences are? |
| 15:55.20 | danfromuk | Sorry, i meant upgrade.txt :-S |
| 15:55.24 | [TK]D-Fender | Ice_Strike: If YOUR configured your dialplan... what do you think the SAMPLE one is going to do for you? |
| 15:56.11 | davlefouAMD | [TK]D-Fender, yes, it that i look for solve it! |
| 15:56.13 | gerritfromsa | Ice_Strike,: If you had any priority jumping that's gone |
| 15:57.26 | gerritfromsa | Ice_Strike, also watch for VARIABLES , I remember CALLERIDNUM in 1.2 changed to CALLERID(num) in 1.4 etc... |
| 15:57.29 | danilom | i dont know was wrong but i delete the manager.conf and create a new one, same config and work |
| 15:57.41 | Ice_Strike | [TK]D-Fender Usally I have changed sip.conf extensions.conf queues.conf agents.conf manager.conf amd.conf |
| 15:57.45 | [TK]D-Fender | gerritfromsa: this wasn't a dialplan-exclusive question... |
| 15:57.54 | Greenlight | danilom: Funky character or whitespace. |
| 15:57.54 | [TK]D-Fender | gerritfromsa: You should read what you're answering to... |
| 15:58.06 | gerritfromsa | That could be windows , next time use dos2unix in bash to convert |
| 15:58.18 | Greenlight | Naa, he used nano |
| 15:58.34 | [TK]D-Fender | Not read... just not reading... |
| 15:59.22 | danfromuk | danilom: maybe a permissions issue? A new file could have the correct permissions? |
| 15:59.55 | gerritfromsa | side affect of multi tasking |
| 16:00.10 | danilom | danfromuk, well i check the perm and was ok |
| 16:00.50 | Ice_Strike | Luckaly I don't use priorityjumping |
| 16:01.07 | Ice_Strike | I see SetMusicOnHold(default) wont work accordign to changes. |
| 16:01.36 | gerritfromsa | Yes and the astdb stuff |
| 16:02.50 | Ice_Strike | asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory |
| 16:02.53 | Ice_Strike | damn |
| 16:03.24 | gerritfromsa | I love this script to install dependencies : asterisk_source_dir/contrib/scripts/install_prereq install |
| 16:03.44 | gerritfromsa | Ice_Strike, try it , it works great to fix that |
| 16:03.54 | gerritfromsa | The script I mean |
| 16:04.06 | Ice_Strike | I did use install_prereq ;) |
| 16:04.34 | gerritfromsa | check with #rpm -qa | grep ssl |
| 16:04.50 | gerritfromsa | you should get openssl etc |
| 16:05.00 | gerritfromsa | or configure --without it |
| 16:05.38 | gerritfromsa | Ice_Strike, sorry often I just go of without reading ... |
| 16:05.56 | Ice_Strike | yea I get openssl etc |
| 16:06.10 | gerritfromsa | Ice_Strike, what OS? |
| 16:06.27 | gerritfromsa | and openssl-devel ? |
| 16:06.46 | Ice_Strike | centos |
| 16:07.04 | Ice_Strike | https://issues.asterisk.org/jira/browse/ASTERISK-20407?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel |
| 16:08.41 | gerritfromsa | where did you put the source code - /usr/src/ ? |
| 16:09.11 | Ice_Strike | ldconfig |
| 16:09.16 | Ice_Strike | got it to work now |
| 16:09.27 | Ice_Strike | you need to refresh your dynamic linker cache using ldconfig |
| 16:09.45 | gerritfromsa | shot |
| 16:10.56 | gerritfromsa | Ice_Strike, Default codec for IAX2 changed fro gsm to g729 |
| 16:11.42 | gerritfromsa | Luckily open source g729 works great |
| 16:11.59 | Greenlight | Umm |
| 16:12.07 | Greenlight | There is no "open source g729" |
| 16:12.25 | gerritfromsa | a free one then - with no limitations |
| 16:12.31 | Greenlight | There's an unlicenced version |
| 16:12.45 | Greenlight | Which steals from the copyright holders. |
| 16:12.49 | gerritfromsa | depends on your country's laws |
| 16:13.14 | Greenlight | Either way, best to avoid mention of it in here. |
| 16:13.15 | Ice_Strike | Damn, sip show peers no longer work |
| 16:13.15 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 16:13.18 | gerritfromsa | a codec is a mathematical formula , copyright is the thief here |
| 16:13.23 | Ice_Strike | No such command 'sip show peers' (type 'core show help sip show' for other possible commands) |
| 16:13.28 | Greenlight | This being a digium sponsored channel and all. |
| 16:13.56 | gerritfromsa | even digium makes mistakes |
| 16:14.02 | Ice_Strike | No such command 'reload' |
| 16:14.04 | Ice_Strike | lol |
| 16:14.14 | navaismo | module lad <modulename> |
| 16:14.25 | navaismo | s/lad/load/ |
| 16:14.27 | Greenlight | Ice_Strike: Sounds like chan_sip isn't loaded |
| 16:14.36 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:14.36 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:14.45 | gerritfromsa | or no sip.conf in /etc/asterisk |
| 16:15.00 | gerritfromsa | make samples ... |
| 16:15.03 | Ice_Strike | there is |
| 16:15.24 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:15.41 | wdoekes | Ice_Strike: module load chan_sip.so |
| 16:15.47 | wdoekes | it should report an error somewhere |
| 16:15.54 | gerritfromsa | recompile |
| 16:16.13 | Ice_Strike | <PROTECTED> |
| 16:16.18 | Ice_Strike | Haa |
| 16:16.21 | Ice_Strike | Aha |
| 16:17.09 | Ice_Strike | I thought it will be installed by default. |
| 16:18.24 | [TK]D-Fender | What is "default"? |
| 16:18.51 | Ice_Strike | Sip. I am recompiling now. |
| 16:23.44 | *** join/#asterisk charkee (~charkee@122.55.36.17) |
| 16:24.39 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 16:28.14 | *** join/#asterisk nix8n82 (~AndChat27@2600:100e:b001:332d:c951:ca24:617b:b592) |
| 16:33.06 | Ice_Strike | "dialplan reload" did not show error what applications have been Removed or Deprecated |
| 16:33.24 | Ice_Strike | I thought it would. |
| 16:33.30 | Greenlight | If only such changes were documented in a file somewhere |
| 16:34.09 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.244.113) |
| 16:34.24 | davlefouAMD | i try it an new time but nothing. I get src then ilbc but i have the same message. |
| 16:35.00 | davlefouAMD | it can be not the same |
| 16:35.28 | [TK]D-Fender | Ice_Strike: You can grep that out yourself, but * will not check on load, it'll only fail in run-time |
| 16:35.47 | [TK]D-Fender | Ice_Strike: Also parameters have changed on some |
| 16:39.44 | Ice_Strike | Yep |
| 16:41.58 | ghost75 | somebody was calling my mobile and i am using chan_mobile, so it goes to *. Why is this number showing as "fake auth rejection" |
| 16:42.31 | Ice_Strike | Aha AgentCallbackLogin() removed. |
| 16:43.36 | navaismo | you can read the chagelog txt |
| 16:43.46 | ghost75 | so it looks like a hacker but only somebody called me oO |
| 16:43.56 | navaismo | startinga at 1.6 then 1.8 then 10 then 11 |
| 16:55.43 | davlefouAMD | if i have code time problem, how can i find code time? |
| 16:56.50 | [TK]D-Fender | ghost75: Show us |
| 16:59.34 | ghost75 | http://pastebin.com/9WdKgLCu |
| 17:00.18 | *** join/#asterisk WcktKlwn (~WcktKlwn@c-24-9-36-149.hsd1.co.comcast.net) |
| 17:00.35 | ghost75 | 015155059373 is the nr who called me and shows as fake auth |
| 17:05.53 | *** join/#asterisk zigg (~matt@unaffiliated/zigg) |
| 17:10.02 | [TK]D-Fender | ghost75: that fake auth has nothing to do with chan_mobile |
| 17:10.21 | [TK]D-Fender | [Oct 29 16:25:58] NOTICE[27672] chan_sip.c: Sending fake auth rejection for device "015155059373" <sip:015155059373@arcor.de;user=phone>;tag=0002bd48 |
| 17:10.22 | [TK]D-Fender | [Oct 29 16:26:25] VERBOSE[29019] pbx.c: -- Executing [s@incoming-mobile:1] Goto("Mobile/alcatel-a077", "08192998780,1") in new stack |
| 17:10.33 | ghost75 | that number called me on mobile and went to * over chan_mobile |
| 17:10.34 | [TK]D-Fender | ghost75: half a minute apart.. and that auth is a SIP thing, not chan_mobile. |
| 17:10.40 | [TK]D-Fender | that was a SIP CALL |
| 17:10.41 | [TK]D-Fender | ^ |
| 17:10.54 | [TK]D-Fender | [Oct 29 16:25:58] NOTICE[27672] chan_sip.c: Sending fake auth rejection for device "015155059373" <sip:015155059373@arcor.de;user=phone>;tag=0002bd48 |
| 17:11.00 | [TK]D-Fender | ardor.de <---------------- |
| 17:11.12 | [TK]D-Fender | this is NOT a cellular call with chan_mobile |
| 17:11.15 | ghost75 | why is it shown as fake auth |
| 17:11.46 | [TK]D-Fender | because it is a basic challenge to whatever comm they are actually sending you |
| 17:13.57 | ghost75 | in cdr the number doesnt appear |
| 17:15.24 | ghost75 | so maybe with chan_mobile somethings shows up as sip devices |
| 17:16.18 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.244.113) |
| 17:17.22 | [TK]D-Fender | no |
| 17:17.34 | [TK]D-Fender | they are 2 completely, ABSOLUTELY seperate calls |
| 17:17.39 | [TK]D-Fender | they have NOTHING to do with one another |
| 17:17.50 | [TK]D-Fender | look at the time-stamps |
| 17:18.11 | [TK]D-Fender | And you aren't even paying attention to SIP debug and you';re wondering what the challenge is to... |
| 17:19.03 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 17:19.12 | ghost75 | so whatever it is but this fake auth thing confuses me, because this looks like hacking |
| 17:19.44 | ghost75 | sip debug was not on during that time |
| 17:20.54 | [TK]D-Fender | There is no valid detail to base that on |
| 17:21.14 | [TK]D-Fender | You can't tell a SIP comm from that of a completely different channel-type.... |
| 17:25.14 | *** join/#asterisk nix8n82 (~AndChat27@2600:100e:b001:332d:c951:ca24:617b:b592) |
| 17:33.12 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 17:38.28 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 17:40.29 | Ice_Strike | Been reading though changes logs, Look like Macro, SetCallerID, ResponseTimeout, SetMusicOnHold, AgentCallbackLogin removed |
| 17:40.35 | Ice_Strike | need to update my dial plan |
| 17:43.58 | [TK]D-Fender | half of that should have been changed in 1.2 ... lforget about 1.4+ |
| 17:44.27 | Ice_Strike | I was upgrading from 1.4 to 11 |
| 17:44.57 | [TK]D-Fender | yes, and half of that wasn't what you should have been doing in 1.4 either |
| 17:51.21 | *** join/#asterisk serafie (~erin@nat/digium/x-qqymgnnqfzsirptn) |
| 17:52.19 | *** join/#asterisk nix8n82 (~AndChat27@2600:100e:b001:332d:c951:ca24:617b:b592) |
| 17:52.55 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
| 17:54.30 | *** join/#asterisk vitovt (~vitovt@77.120.244.37) |
| 17:54.54 | vitovt | Hi all! Does anybody can help with Elastix ? |
| 17:55.20 | vitovt | <PROTECTED> |
| 17:56.05 | Penguin | Can you write custom firewall rules inside elastix? |
| 17:56.44 | vitovt | i didn't find such options via gui |
| 18:01.54 | Penguin | If I couldn't find any other way to do it, I would look to see if there are any postconfig commands that can be issued. |
| 18:03.19 | Penguin | If there isn't that, and if you make changes in that GUI often, I would write a shell script to run from crontab every minute. The shell script would check to see if the MASQUARADE exists in iptables; if it does, exit; if it does not, add it. |
| 18:12.50 | Ice_Strike | Does Macro(agent_exten,${EXTEN:1},20) need to change to Gosub? |
| 18:16.33 | vitovt | ok thanks |
| 18:21.33 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 18:22.47 | [TK]D-Fender | Ice_Strike: no |
| 18:23.05 | [TK]D-Fender | Ice_Strike: "core show application macro" should answer that for you |
| 18:23.17 | [TK]D-Fender | Ice_Strike: is it THERE? have the parameters changed? |
| 18:24.37 | *** join/#asterisk nix8n82 (~AndChat27@2600:100e:b001:332d:c951:ca24:617b:b592) |
| 18:28.22 | atha | vitovt MASQUERADE |
| 18:30.16 | vitovt | atha MASQUARADE disappears if I add it to iptables rules manually |
| 18:30.23 | vitovt | how to add it via gui ? |
| 18:30.41 | vitovt | or change gui apply script |
| 18:30.56 | vitovt | beacause it deletes *nat portion from config |
| 18:31.25 | Penguin | I don't know the first thing about elastix, so I would modify it or work around it. |
| 18:31.34 | [TK]D-Fender | #elastix <------- |
| 18:31.57 | vitovt | tnx, but there is no respond in #elastix ... |
| 18:32.23 | [TK]D-Fender | Well this is a distro issue, not an Asterisk issue |
| 18:32.39 | [TK]D-Fender | If their stuff is specifically in your way then it's up to them |
| 18:32.53 | atha | vitovt MASQUERADE |
| 18:32.55 | atha | ;D |
| 18:33.18 | atha | s/MASQUARADE/MASQUERADE/g |
| 18:33.19 | Penguin | Perhaps his iptables is in a different language. |
| 18:37.07 | *** join/#asterisk serafie (~erin@nat/digium/x-fqxpnvcwmtqewlcm) |
| 18:44.49 | *** part/#asterisk opticron (~opticron@pianoben.ch) |
| 18:49.27 | gerritfromsa | vitovt, I believe elastix has a sip_custom.conf - if you add accounts there the gui doen not change them |
| 18:49.49 | Penguin | That isn't relevant. |
| 18:51.03 | gerritfromsa | "I don't know the first thing about elastix, so I would modify it or work around it" |
| 18:51.30 | gerritfromsa | That's irrelevant |
| 18:51.34 | Penguin | That's what I said, yeah. |
| 18:52.44 | Penguin | I will probably never use elastix. Ever. |
| 18:53.02 | Penguin | I have no desire to nor a need to. |
| 18:53.14 | gerritfromsa | likewise , none of the gui stuff really works |
| 18:53.34 | Penguin | It works enough to ruin that guy's iptables rule that he entered manually, though. |
| 18:53.59 | [TK]D-Fender | [14:30]vitovtatha MASQUARADE disappears if I add it to iptables rules manually |
| 18:54.01 | gerritfromsa | but i guess cli and conf files is not for everyone |
| 18:54.05 | [TK]D-Fender | This doesn't say when things happen |
| 18:54.08 | [TK]D-Fender | Or how he added it |
| 18:54.16 | [TK]D-Fender | Or what he did after |
| 18:55.20 | Penguin | He said he added the rule manually in iptables and he said when he reconfigures in elastix, the rule disappears. |
| 18:55.22 | gerritfromsa | that put him off properly |
| 18:55.24 | [TK]D-Fender | I be interested in seeing how this sip_custom.conf (pretty sure that isn't an actual file) would have any impact on a completely different non-Asterisk related application |
| 18:56.05 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.244.113) |
| 18:56.18 | gerritfromsa | he said the nat portion of his sip.conf disappears |
| 18:56.30 | Penguin | That's not at all what he said. |
| 18:56.36 | [TK]D-Fender | sip.conf is always blown away regardless. |
| 18:56.50 | [TK]D-Fender | He's GUI'd You NEVER touch sip.conf on a system like that |
| 18:56.58 | Penguin | When English isn't the first, or even the second, language, sometimes the details don't appear to be as clear. |
| 18:57.52 | gerritfromsa | elastix , trixbox etc allows you to change conf but only in the _cutom.conf files |
| 18:57.59 | gerritfromsa | the rest is overwritten by the gui |
| 18:58.00 | [TK]D-Fender | Well aware |
| 18:58.08 | [TK]D-Fender | So anything he does in sip.conf = suicide |
| 18:58.18 | gerritfromsa | thats it |
| 18:58.22 | Penguin | I doubt ANY of those distros care about the iptables nat table. |
| 18:59.05 | gerritfromsa | and some of the users here only speak one language |
| 18:59.11 | [TK]D-Fender | [13:55]vitovtI wanna use elastix machine like network gateway. I add MASQUARADE rule to iptables config but after reconfigure firewall via elastix gui *nat section disappears |
| 18:59.26 | [TK]D-Fender | I'd be wondering more if it's because he now has a PUBLIC interface... |
| 19:00.32 | gerritfromsa | it probably uses shorewall and the gui make the iptable changes via shorewall |
| 19:01.08 | gerritfromsa | if his interfaces is not set up just right it will ignore the masq |
| 19:04.28 | gerritfromsa | quit |
| 19:05.03 | gerritfromsa | EXIT |
| 19:05.11 | navaismo | kick |
| 19:05.42 | gerritfromsa | how does one quit this channel via a command |
| 19:05.49 | [TK]D-Fender | wow |
| 19:06.16 | [TK]D-Fender | <PROTECTED> |
| 19:08.18 | drmessano | rm -... oh nevermind |
| 19:08.31 | [TK]D-Fender | Full of so much flat-out wrong information, poor techniques & methodologies ..... and also doesn't know enough basic IRC to even LEAVE on his own... |
| 19:08.39 | *** join/#asterisk gerritfromsa (~gerritfro@8ta-229-226-173.telkomadsl.co.za) |
| 19:08.53 | [TK]D-Fender | drmessano: GO GO GO!!!! |
| 19:08.55 | [TK]D-Fender | awww |
| 19:09.30 | [TK]D-Fender | And when he quits he can't even stay successfully quit ;) |
| 19:09.55 | drmessano | Maybe he's only ever been banned |
| 19:10.14 | drmessano | "To exit a channel I just keep cursing until the scroll stops" |
| 19:10.19 | [TK]D-Fender | LOL |
| 19:10.59 | drmessano | That's how I rolled on IRC from 97 to the early 2000s. Just sayin |
| 19:12.08 | jeev | drmessano, you must've been on DALnet |
| 19:12.17 | drmessano | Undernet and Newnet |
| 19:12.27 | *** join/#asterisk gerritfromsa (~gerritfro@8ta-229-226-173.telkomadsl.co.za) |
| 19:12.33 | jeev | haha |
| 19:12.41 | drmessano | Undernet until it buckled under attacks... then Newnet until it split up |
| 19:12.42 | jeev | i think i linked a server to newnet once |
| 19:12.48 | jeev | no efnet ? |
| 19:13.10 | drmessano | Nope. Efnet was only good for Warez and cybersex |
| 19:13.12 | gerritfromsa | figured out the /help stuff now |
| 19:13.22 | jeev | drmessano, not really |
| 19:13.29 | drmessano | Yeah really |
| 19:13.31 | gerritfromsa | thanks |
| 19:13.56 | drmessano | Oh and trivia bots with rigged games |
| 19:14.28 | drmessano | Nearly every single one had a cheat script or favored the OPs. |
| 19:15.25 | jeev | i used to own shit on efnet |
| 19:15.30 | drmessano | Dalnet was just a haven for trolls. They should have called it "Under the Bridge" |
| 19:16.21 | drmessano | So you linked a server on Newnet, possibly |
| 19:16.26 | drmessano | Were you like 15 at the time? |
| 19:16.58 | drmessano | I knew a kid named Jeev on there that was like 15ish back in 2001 maybe |
| 19:17.13 | jeev | yea except it was naptime back then not jeev. |
| 19:17.18 | drmessano | oh ok |
| 19:19.04 | drmessano | I ran several trivia channels. I maintained one of the more popular trivia bot scripts |
| 19:19.10 | drmessano | Good times, mostly |
| 19:26.14 | jeev | trivia was fun |
| 19:26.24 | jeev | but i was more in the business of taking over #shells and messing with the big boys. |
| 19:26.29 | jeev | netsplit.. |
| 19:28.31 | drmessano | Oh, so you were a script kiddie? |
| 19:28.42 | drmessano | Can't say it surprises me |
| 19:28.47 | drmessano | :D |
| 19:32.19 | jeev | no i wasn't a script kiddie |
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| 20:06.44 | scgm11 | hi is it possible to have a carriage return (\r\n) on a variable in the dial plan? |
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| 21:15.17 | *** join/#asterisk DeadM0nkey (~brian@50-79-249-253-static.hfc.comcastbusiness.net) |
| 21:16.19 | DeadM0nkey | Compared to SIP/RTP, is IAX more robust in terms of handling latency? |
| 21:16.34 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 21:16.49 | scgm11 | hi is it possible to have a carriage return (\r\n) on a variable in the dial plan? |
| 21:17.59 | [TK]D-Fender | DeadM0nkey: Nope |
| 21:19.20 | *** join/#asterisk threesome (~threesome@94.113.13.143) |
| 21:21.32 | DeadM0nkey | [TK]D-Fender some of our clients, when leaving our facility choose to continue using our phone service. Currently we set them up with a firewall and point to point VPN. I was pondering if there were any advantages to deplying a small asterisk box at their location to cut down on any potential call quality issues. Sounds like there would be no advantage. Thoughts? |
| 21:21.50 | *** part/#asterisk scgm11 (~Sebastian@r190-132-155-50.pocancel.mobile.ancel.net.uy) |
| 21:22.33 | [TK]D-Fender | depends what is going on at each leg and where the traffic is actually going |
| 21:22.59 | Penguin | How would it make any difference if you add another asterisk box in the remote location? You've still got the same network between the same two points. |
| 21:23.43 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 21:24.29 | DeadM0nkey | Because I read about IAX being more efficient than sip it would allow the phones to talk the the local box, then the local box to speak to the main cluster. However if IAX is no more robust than SIP/RTP, it sounds like the idea is a dud |
| 21:25.58 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 21:26.51 | [TK]D-Fender | how is the path any better for putting that in between? |
| 21:27.30 | [TK]D-Fender | There might be other reasons to do thi, but quality end-to-end isn't it |
| 21:28.46 | DeadM0nkey | Thanks |
| 21:30.50 | *** join/#asterisk Ice_Strike (~Ice_Black@84.92.51.164) |
| 21:34.39 | Natureshadow | Hmm, I am trying to do CLIR via SIP on a PSTN gateway. I have tried several combinations of the Privacy, P-Asserted-Identity and Remote-Party-ID headers, to no avail. |
| 21:34.47 | Natureshadow | What's the correct way to do that? |
| 21:34.59 | *** join/#asterisk raub (~raub@ip70-171-22-39.ga.at.cox.net) |
| 21:36.24 | Ice_Strike | Where the link to get more infor what does: dialplan add extension do |
| 21:36.39 | gartral | this is very strange.. I'm having a ptsn<->sip issue.. i think, If i recieve calls from a regular phone, everything works fine, if i make or recive sip calls, fine, but if I dial out, the other party can't hear me.. any thought? |
| 21:39.20 | gartral | stupid router, think i fixed it! |
| 21:40.11 | Ice_Strike | ~books |
| 21:40.15 | Ice_Strike | ~book |
| 21:40.15 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:45.57 | kaldemar | Ice_Strike: core show help dialplan add extension |
| 21:46.14 | dwayne | Qwell, is there a recommended way to enable SIP TLS / SRTP on AsteriskNOW 3.0 ? I imagine the only option is to build res_srtp and rebuild asterisk |
| 21:46.22 | Ice_Strike | thank you |
| 21:49.02 | Ice_Strike | I am getting No such command 'save dialplan' in asterisk 11 |
| 21:49.12 | Ice_Strike | It work fine in Asterisk 1.4 |
| 21:49.21 | Ice_Strike | so its dialplan save? |
| 21:50.08 | [TK]D-Fender | Why would you be doing this in the first place? |
| 21:51.26 | Ice_Strike | [TK]D-Fender After dialplan remove extension command, then I use dialplan save |
| 21:52.53 | [TK]D-Fender | Ice_Strike: why are you attempting to manipulate the dialplan at all from CLI? |
| 21:56.17 | Ice_Strike | Unless you know alternative good way. I use "dialplan extension add" to add many campaign as I like. |
| 21:57.11 | [TK]D-Fender | Ice_Strike: perhaps your strategy as a whole is poor. |
| 21:57.23 | *** join/#asterisk felipe_ (~felipe@unaffiliated/felipe) |
| 21:58.01 | [TK]D-Fender | Ice_Strike: Your use of "campaign" is also quite vague |
| 22:01.37 | mjordan | Ice_Strike: Without recommending an approach to your particular issue, lots of CLI commands changed in the great CLI command migration of post-1.4. You may want to look at cli_aliases.conf, which contains many aliases from previous versions of Asterisk to the current version you are installing. You can also use cli_aliases.conf to have the same syntax as previous versions. |
| 22:02.29 | mjordan | In general, I'd agree with [TK]D-Fender. You can almost always find a way to do what you want to do without requiring dynamic dialplan manipulation. |
| 22:03.39 | Ice_Strike | [TK]D-Fender Using web interface, I can add create campaigns, so I can assign agents to specific campaign. When I add a campaign on the web, "dialplan extension add" get executed - eg: Extension '4,1,Macro(outbound_campaign|campaign-queue-a|4)' added into 'outbound_campaigns' context |
| 22:03.49 | [TK]D-Fender | I can only see this being useful in desperate situations where you have no other access |
| 22:04.14 | [TK]D-Fender | Ice_Strike: Already I see you using illegal pipe chars in there |
| 22:04.25 | [TK]D-Fender | Ice_Strike: more bad habits from 1.0 |
| 22:04.39 | [TK]D-Fender | Ice_Strike: And there are typically better overall strategies |
| 22:05.00 | [TK]D-Fender | Ice_Strike: including generating actual config files and reloading |
| 22:05.34 | Ice_Strike | mjordan I see, thanks |
| 22:05.58 | Ice_Strike | [TK]D-Fender Which better strategies do you recommend? |
| 22:07.12 | [TK]D-Fender | Already just gave one |
| 22:07.26 | [TK]D-Fender | Next I'd have to see what is being done more specifically that warrants it |
| 22:09.00 | Ice_Strike | Which actual config files? |
| 22:11.18 | [TK]D-Fender | extensions.conf |
| 22:11.35 | [TK]D-Fender | ]make PHYSICAL files instead of having the dialplan generate the text it its own sloppy way. |
| 22:11.43 | [TK]D-Fender | You want to output them>? make them direct |
| 22:13.47 | Ice_Strike | Hmmmm |
| 22:14.00 | Ice_Strike | I like to see example code strategies, so I get the idea.. |
| 22:17.46 | WIMPy | Ist your extensions.conf still readable when you use dialplan save? |
| 22:17.59 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 22:24.18 | Ice_Strike | [TK]D-Fender Can you provide a bit more info how to generating actual config files and reloading? |
| 22:24.35 | Ice_Strike | Any open source script does that? |
| 22:25.05 | WIMPy | Have you ever written any script? |
| 22:25.32 | Ice_Strike | ofcourse |
| 22:25.34 | WIMPy | Think of your own config file (or database if you prefer) and generate your configs from that. |
| 22:25.48 | WIMPy | then I don't understand the question. |
| 22:32.27 | [TK]D-Fender | Ice_Strike: You need a sample to output a line of text to a file? |
| 22:32.57 | [TK]D-Fender | asterisk -rx "reload" |
| 22:32.59 | [TK]D-Fender | DONE |
| 22:35.09 | snadge | has anyone else used that star trinity sip tester product? |
| 22:35.35 | snadge | i feel like a bit of a noob for cracking it out.. but it has a gui and sliders, and things you can click on it.. and generates graphs, and reports etc |
| 22:36.42 | snadge | if you send 50 calls, 10 at a time, 1 millisecond apart.. to an asterisk server, the call answering time goes from about half a second.. to 3 seconds |
| 22:36.58 | snadge | i found that semi interesting :P |
| 22:37.12 | *** join/#asterisk Pullphinger (~Pullphing@c-24-13-69-42.hsd1.il.comcast.net) |
| 22:37.34 | snadge | that was 2 a cluster of 4 asterisk servers even |
| 22:37.41 | snadge | err s/2/to |
| 22:38.44 | snadge | im the resident asterisk noob at the moment.. hopefully one day i'll have a bit more experience and be slightly less annoying ;) |
| 22:44.47 | dijib | SeRi: home yet? |
| 22:44.56 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 22:45.14 | Preatorian | Imagines Shrek " Are we there yet" ... " Are we there yet?" ... |
| 22:48.31 | snadge | can anyone recommend a migration path from Asterisk to Microsoft Lync? |
| 22:49.09 | snadge | sorry.. my sense of humour is a little bit twisted |
| 22:57.16 | Preatorian | I can, open the window, and throw out asterisk. and start over. Microsoft doens't support it (Trust me, tried, we were quicker simply building from scratch, only pays off if you have like 1000+ phone setups). |
| 22:59.08 | Preatorian | I still prefer asterisk over lync any time when it comes to small setup's/companies (less then 500 phones). |
| 22:59.35 | *** join/#asterisk bchamberlain (~brian98@unaffiliated/brian98) |
| 22:59.43 | Preatorian | But that's 'personal opinion' nothing related on facts or anything :) |
| 23:00.34 | snadge | oh okay.. so Lync might be something someone would look into if they had more than 500 seats |
| 23:00.45 | snadge | i just assumed there was no reason to use it at all ;) |
| 23:02.40 | Preatorian | In larger coorperations it has it's uses, but only if the actual backend is also integrated with microsoft applicatons like office 365/sharepoint bla bla then it helps. for small companies, that equipment is way to expensive (relative speaking). |
| 23:03.42 | Penguin | its uses <---- |
| 23:05.08 | Preatorian | See what i did there ;-) |
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| 23:25.30 | Ice_Strike | [TK]D-Fender sure |
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