14:50.08 | *** join/#asterisk infobot (ibot@69-58-76-73.ut.vivintwireless.net) |
14:50.08 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.4.0 (2015/06/04), 11.18.0 (2015/06/04), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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15:05.43 | reveal | man wimpy im looking at asterisk questions on google you are everywhere hehe |
15:07.02 | reveal | WIMPy: also it never even gets to the point of trying to pass the call on to the PBX when it complains about local permissions |
15:07.05 | WIMPy | Not really. But maybe google likes me. I don't know. |
15:07.11 | reveal | pfft |
15:07.19 | reveal | http://infobot.rikers.org/%23asterisk/20120127.html.gz |
15:07.21 | reveal | LOL |
15:07.29 | WIMPy | That means we're back to: >> So the extension doesn't exist. At leas not in a context you're allowed to use. |
15:07.56 | reveal | now ? where does it not exist on asterisk or my other pbx |
15:09.27 | reveal | im really green to asterisk if you havent figured that out already WIMPy |
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15:16.26 | WIMPy | If asteris tells you you're not allowed to call that extension then that's what it is. |
15:18.10 | WIMPy | But it's time to come up with more details about what you're doing. |
15:19.24 | [TK]D-Fender | revealim really green to asterisk if you havent figured that out already WIMPy <- then you need to read the book and learn how * works. |
15:19.27 | [TK]D-Fender | ~book |
15:19.27 | 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 |
15:19.29 | [TK]D-Fender | ^^^ |
15:19.29 | reveal | ok where do i look to add then in the |
15:19.52 | [TK]D-Fender | extensions.conf <- |
15:20.11 | [TK]D-Fender | Go learn about the dialplan |
15:20.17 | [TK]D-Fender | Dialplan = 90% of Asterisk |
15:21.10 | carrar | SO MUCH LEARN |
15:21.40 | [TK]D-Fender | SUCH KNOWLEDGE |
15:21.42 | [TK]D-Fender | WOW |
15:21.52 | carrar | heh |
15:21.59 | [TK]D-Fender | </shiba> |
15:22.09 | reveal | [TK]D-Fender: thanks heh |
15:22.17 | WIMPy | Or your last chance to live a happy life. |
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15:36.58 | *** join/#asterisk net (79f28b33@gateway/web/freenode/ip.121.242.139.51) |
15:37.26 | net | am using dahdi and how to maintain callerid even i transfer a call? |
15:39.23 | WIMPy | By not changing it. |
15:40.41 | net | WIMPy: am not changing anything, i am just using set ${CALLERID(num)} in a variable and doing attended transfer |
15:40.48 | net | WIMPy: will blind transfer works? |
15:41.05 | WIMPy | How do you not change it when you do that Set? |
15:41.37 | net | WIMPy: am just setting up the default value... am not overriding any value on my own |
15:41.51 | net | WIMPy: the reason is i have to get that in a different context |
15:42.02 | [TK]D-Fender | Attended = gets your CID, not the transferee. Taht's how it works |
15:42.22 | WIMPy | Until the transfer completes. |
15:44.43 | [TK]D-Fender | Even then depends if the tech supports updates |
15:45.22 | WIMPy | That's true for all features. |
15:46.10 | net | WIMPy: thank you |
15:46.13 | net | [TK]D-Fender: thank you |
15:46.46 | net | am doing now a blind transfer without setting up anything, and am able to see the callerid in the NoOp |
15:46.50 | net | thank you |
15:47.51 | *** join/#asterisk raj (~raj@unaffiliated/cypha) |
15:51.08 | reveal | WIMPy OSDIAL is managing asterisk, and all the tools being used such as mysql and apache, osdial generates almost all the config files asterisk uses, to allow for future changes to asterisk and leave some flexibility for others to make their own modification that isnt generated by osdial, so osdial generates the phones and carriers |
15:51.28 | reveal | so that means i need to deal with the people that support the osdial |
15:51.48 | [TK]D-Fender | if you can't mod it cleanly.... then it is what it is |
15:51.58 | reveal | yes |
15:52.25 | [TK]D-Fender | I'd check if you actually did their GUI part properly |
15:52.27 | [TK]D-Fender | a few times |
15:53.13 | reveal | it cane make and receive calls just problem with internal dialing before it hits the pbx and the ext exists within osdial that passes it to asterisk |
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16:03.32 | mdhas | Question is asterisk/freepbx a CPU intensive application or is a Memory intensive application or is neither of those statements true |
16:03.55 | [TK]D-Fender | too vague |
16:04.03 | WIMPy | Depends on what you want to do with it. |
16:04.09 | mjordan | And the number of calls. |
16:04.10 | [TK]D-Fender | Depends entirely on usage |
16:04.22 | WIMPy | But people use it on Raspberry Pis/ |
16:04.33 | mdhas | D-Fender: Let's say an ordinary office usage, extensions calling extensions, sales calling out...about 100 people... |
16:04.38 | [TK]D-Fender | Many people also deploy large clusters of servers |
16:04.44 | [TK]D-Fender | Your needs might fall.... anywhere |
16:05.22 | [TK]D-Fender | mdhas: Anything special there? Transcoding? recording? external processing? |
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16:12.01 | BigKristoff | Hi, does anyone know something abouth MOH RealTime not woring in asterisk 11 ? |
16:12.07 | BigKristoff | working* |
16:12.27 | BigKristoff | My asterisk does not see real Time Moh Classes |
16:12.48 | [TK]D-Fender | Show what you've got. |
16:17.31 | BigKristoff | what you mean ? |
16:19.55 | [TK]D-Fender | Configs, databs and CLI status dumps |
16:22.00 | BigKristoff | extconfig.conf -> musiconhold => odbc,asterisk,moh (my whole Asterisk configuration is real time - queues, peers etc. and they work fine) |
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16:23.52 | BigKristoff | CLI> moh show classes |
16:23.58 | BigKristoff | epmty |
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16:24.54 | BigKristoff | musiconhold.conf: [general] |
16:24.54 | BigKristoff | ;cachertclasses=yes <- commented, default no |
16:25.00 | BigKristoff | [default] |
16:25.01 | BigKristoff | mode=files |
16:25.01 | BigKristoff | directory=moh |
16:25.55 | BigKristoff | and DB schema is standard for real time conf for moh |
16:30.46 | [TK]D-Fender | pastebin it all |
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17:45.53 | monsterco | anyone know the default yealink phone set password? not the webui pass |
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18:47.56 | Cuzner | probably a stupid question, but can I do multiple Set()'s on a single line? |
18:48.16 | WIMPy | Array() |
18:50.34 | Cuzner | WIMPy: I'd love to use an array, but i need to set values for existing declared vars. |
18:51.17 | Cuzner | it's no biggie, i can do it on two lines, i'm just always looking for the 1-line approach :) |
18:51.37 | WIMPy | Asterisk doesn't know arrays. |
18:52.17 | Cuzner | ohhh |
18:52.18 | Cuzner | i see |
18:52.26 | Cuzner | yeah, this is exactly what i want then, thanks |
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19:45.37 | dan_j | Hi. Do Queues have a H extension for calls that are not answered from the queue? |
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19:54.11 | [TK]D-Fender | dan_j: when is "not answered" determined? |
19:55.37 | dan_j | When the caller hangs up while they are in the queue. Queue timeout is 999 seconds which never gets hit. |
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19:57.57 | [TK]D-Fender | If they hangup while in queue, that call has it's own access to H like everything else |
19:57.57 | Penguin | [TK]D-Fender: its |
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20:03.21 | dan_j | Which context is it in? The one that called the Queue program? |
20:06.35 | [TK]D-Fender | yes, just like always |
20:06.39 | [TK]D-Fender | that call is just like any other |
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20:18.41 | dan_j | Ok. Thanks for the confirmation. |
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20:53.49 | pjensen00 | mjordan: In ARI I'm doing two snooop channels that are spying on a single channel. I'm trying to record the "in" and the "out" audio streams. I'm getting them to successfully start recording, but both call recordings end up at 44k and no audio in them. |
20:54.53 | pjensen00 | Do you have to put them into a bridge with that same channel? |
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20:57.17 | pjensen00 | CURSES! |
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21:32.44 | Hsilamot | anyone here could help me? i can't make any call with a DAHDI card, only when i pick up a parallel phone the DTMF works |
21:33.22 | WIMPy | Analog is evil! |
21:33.49 | Hsilamot | indeed |
21:33.53 | Hsilamot | very much |
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21:34.03 | Hsilamot | <PROTECTED> |
21:34.28 | Hsilamot | if i let it do by itself the phone company issues a "incomplete number" alert |
21:34.41 | Hsilamot | if i pickup the phone to listen to the DTMF tones myself, it goes all straight |
21:35.19 | nny | I am pretty much a hardcore Vanilla Asterisk kind of guy. That having been said I have a client that wants a FreePBX box. I usually just do maintenance and repair to them. I know this is a loaded question but anyone have a suggestion on the Distro vs. CentOS + Tarball (+ asterisk etc)? I am asking in the freepbx channel but I get all my best advice |
21:35.20 | nny | <PROTECTED> |
21:35.58 | WIMPy | Their distro? |
21:36.02 | nny | yeah |
21:36.15 | nny | looks like asterisknow/freepbx + asterisk 11 or 13 |
21:36.17 | WIMPy | At least you know who to blame. |
21:36.21 | nny | and dahdi etc |
21:36.22 | nny | hahaha |
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21:38.30 | nny | oh man glad [TK]D-Fender missed that question, he'd kill me.. er hi! |
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21:40.34 | nny | looks like the distro uses centos anyways, should be fine. |
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22:48.32 | Simon-- | is there a variable or something to disable automatic iax transfer for a particular call? eg, one doing dial(,,g) where we expect the remote end to hang up after something and we want to do more things with the call |
22:50.15 | WIMPy | Sounds l ike you want transfer=mediaonly, but it looks like that's actually broken. |
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23:21.22 | Simon-- | yah, otherwise, it's optimizing away my dialplan |
23:21.35 | Simon-- | I can disable transfer entirely, but I don't want that either |
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23:27.20 | snadge | can someone remind me of a sip call flow? .. invite packet .. response trying.. response ringing.. when the phone is answered, what is sent by the client? |
23:27.33 | K0HAX | 200 OK |
23:27.53 | snadge | it seems im dealing with a stupid router that has some kind of intrusion detection feature which is blocking sip packets |
23:28.08 | snadge | yeah.. we're not getting the 200 OK.. then no response to any sip traffic for several minutes.. or until the router is restarted |
23:28.12 | K0HAX | SIP ALG is your likely culprit |
23:28.26 | K0HAX | that or your NAT timers |
23:28.35 | K0HAX | does the phone actually ring? |
23:28.40 | K0HAX | inbound or outbound? |
23:28.47 | snadge | yes.. and the bug is reliably triggered when the phone is answered |
23:28.54 | K0HAX | weird |
23:29.12 | snadge | but not always.. its strange.. there are reports on the internet that this specific router, has issues with "intrusion detection" and SIP specifically |
23:29.18 | K0HAX | normally with SIP ALG problems (in my experience) the phone never rings, and the SBC sends a bunch of invites that never get answered |
23:29.45 | snadge | right.. we've told the customer to replace their router.. and they have another one, but they've moved house and lost some of the cabling for it :| |
23:29.58 | K0HAX | lol |
23:31.05 | K0HAX | I have a Cisco 2851, I highly recommend it. :P |
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