| 00:00.23 | WIMPy | Officially yes. |
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| 00:35.46 | DBordello | Would you recommend a newbie to start with a front-end gui such as FreePBX, or compile from source and manually edit the config files? |
| 00:35.55 | DBordello | I am not afraid of a CLI and a few config files |
| 00:37.11 | robert_ | [TK]D-Fender: okay okay, hold on and I'll give you a log of what goes on in the console. |
| 00:39.47 | jmetro | DBordello: start from CLI, the gui will gimp your ability to work later. |
| 00:39.56 | WIMPy | DBordello: A frontend is no good if you want to evolve beyond that. It's an easy start but more like a dead end. |
| 00:40.20 | DBordello | I don't plan on doing anything complicated. I am sure the GUI would be sufficient. This is just for home. |
| 00:40.32 | DBordello | but I am worried it will cripple my understand of what is going on |
| 00:40.51 | WIMPy | They are know to be good at that. |
| 00:41.15 | jmetro | Its funny how projects for home can wind up turning complicated when you learn what awesome things you can od. |
| 00:41.26 | WIMPy | But configuring Asterisk is neither easy nor fast. |
| 00:41.27 | DBordello | :) |
| 00:41.37 | jmetro | WIMPy: Depends. For me it was both of those. |
| 00:42.04 | WIMPy | Depends pn what you intend to use it for, sure. |
| 00:42.24 | jmetro | Im only running into trouble now because i'm trying to do this complicated stuff for findme/follow me with/without announcements |
| 00:49.53 | *** join/#asterisk esaym (~esaym153@216-45-91-132.gvec.net) |
| 00:50.06 | esaym | my astcanary doesn't ever start. Anyone know how to debug? |
| 00:51.39 | esaym | 1.8.13 from debian. Just upgraded from 1.8.12 from the digium deb package |
| 00:51.58 | jmetro | Uhhh |
| 00:52.33 | jmetro | [2013-05-29 19:52:01] WARNING[29470]: chan_sip.c:4403 __sip_autodestruct: Autodestruct on dialog 'channel@ip' with owner mypbx in place (Method: BYE). Rescheduling destruction for 10000 ms |
| 00:52.35 | jmetro | spamming |
| 00:52.40 | jmetro | wat are this |
| 01:06.32 | igcewieling | jmetro: you have direct media enabled, that is normal |
| 01:07.00 | robert_ | sup igcewieling |
| 01:07.13 | igcewieling | hello robert_ |
| 01:07.17 | robert_ | hi |
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| 01:12.17 | robert_ | igcewieling: http://pastebin.com/b60zTY0q |
| 01:12.26 | robert_ | actuuaally from earlier, asterisk crashed. |
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| 01:13.32 | robert_ | actually ** |
| 01:17.36 | jmetro | Quebec for Q was probably not the best choice |
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| 01:21.19 | robert_ | igcewieling: actually, should I post a new log? |
| 01:21.55 | robert_ | [TK]D-Fender: there's your log btw. Asterisk's issues with mysql odbc turned into a crash. |
| 01:31.23 | DBordello | Any (free) android soft phone recommendations? |
| 01:33.34 | jmetro | 3cx |
| 01:35.17 | DBordello | thanks, i'll check it out |
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| 01:41.46 | navaismo | linphone |
| 01:43.21 | DBordello | I just noticed I am using 1.8.13. Is this grossly out of date? Or okay to start with? |
| 01:44.18 | jmetro | 1.8 is cool but asterisk 11 is neato |
| 01:44.19 | hebber | I still think 1.8 is on a LTS |
| 01:44.44 | igcewieling | DBordello: upgrade to the latest 1.8 |
| 01:44.55 | WIMPy | 1.8 is LTS, but that version is even out of date for 1.8. |
| 01:44.59 | jmetro | [or 11 because its cooler] |
| 01:45.10 | DBordello | Hmmm, I am using a raspberry pi. Stupid repository is using 1.8.13 |
| 01:45.12 | WIMPy | And off course you miss out on all the new features. |
| 01:45.15 | jmetro | join the force_rport,comedia crowd |
| 01:45.29 | WIMPy | Buid your own 11.4. |
| 01:46.20 | DBordello | I guess building from source isn't a bad plan |
| 01:46.26 | DBordello | finds a cross compiling environment |
| 01:47.05 | igcewieling | DBordello: your hardware makes it more of a hassle, but still worth doing |
| 01:47.24 | DBordello | igcewieling, understandable |
| 01:47.48 | WIMPy | Ou you could do it on the PI. Easier but slower. |
| 01:48.13 | DBordello | WIMPy, we might do that actually |
| 01:48.24 | navaismo | use sdcc |
| 01:48.41 | jmetro | your raspberry pi reminded me of this http://tinyurl.com/9exfwvk |
| 01:48.46 | jmetro | the term "6 year old lego specialist" still gets me |
| 01:49.02 | navaismo | having a bad time printinf out with JAVA HEADACHE!!!! |
| 01:52.32 | DBordello | ugh, I think cross-compiling is harder than just doing it on the hardware |
| 01:54.01 | jmetro | less effort, more time |
| 01:54.04 | jmetro | more effort, cooler result. |
| 01:54.13 | jmetro | but probably still more time ] |
| 01:55.14 | DBordello | :) |
| 02:06.52 | navaismo | i use a screen to compile in the pi and then im going to play eat sleep and then back to see what happened, compiling asterisk take like 2~3hrs |
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| 02:30.18 | DBordello | navaismo, I am attempting to setup a cross-compile environoment now |
| 02:33.42 | apb1963 | start the pi compile; work on the cross compile simultaneously... see which finishes first |
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| 03:16.18 | DBordello | apb1963, that is what I am doing now. I think I have the cross compile setup |
| 03:16.23 | DBordello | but I don't fully understand it |
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| 03:46.21 | esaym | `/part |
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| 04:49.02 | fling | How to properly integrate with sugarcrm? |
| 04:49.11 | fling | I want o have a button for calling. |
| 04:49.46 | fling | Manager uses the button with phone number on the site and asterisk dialing bot this number and manager's number |
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| 09:23.54 | fredericve | Hi, I have a setup with: hylafax <---> asterisk 1.8 with T.38 gateway patch <---> sangoma netborder E1 pri <---> pstn. Between asterisk and netborder is t38 traffic. Faxes randomly fail. with "udptl debug on" I see the udptl packets flowing through asterisk and "fax show sessions" correctly shows the session and I can see the session details as well. I've come to a point where I'm at a loss on how to continue. Anyone has any suggestions |
| 09:23.54 | fredericve | on what to try? |
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| 09:33.32 | bulkorok | fredericve: why do you use hylafax? what does what asterisk rxfax and txfax can't do!? |
| 09:33.48 | fredericve | The results are the same when I replace hylafax with a T38 enabled ATA. At that point asterisk will not start the T.38 gateway |
| 09:34.19 | bulkorok | fredericve: so... again... you can use asterisk self to receive the faxes... |
| 09:34.51 | bulkorok | but I don't know if that is an option in your setupo |
| 09:35.20 | fredericve | bulkorok: for historical reasons. we've been using it since asterisk 1.0 |
| 09:35.44 | bulkorok | fredericve: same here... but I managed to use asterisk for receiving |
| 09:35.51 | bulkorok | reliable... |
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| 09:36.31 | bulkorok | and afaik asterisk 1.8 doesn't need a patch for t38 rxfax and txfax... |
| 09:38.11 | fredericve | thanks for your suggestions, but I can't implement that live at the customer now =/ |
| 09:38.23 | fredericve | I want to understand why these faxes fail |
| 09:38.41 | bulkorok | then you should pastebin logs |
| 09:38.51 | fredericve | we have another customer where the same setup works fine btw |
| 09:39.11 | fredericve | ok, which logs would you need? |
| 09:39.49 | bulkorok | well... asterisk CLI with fax debug |
| 09:40.13 | bulkorok | but mostly I can "guess" what the problem could be... |
| 09:41.51 | bulkorok | and making a tcpdump trace and checking the udptl/t38 flow helps |
| 09:42.35 | fredericve | well i have a pcap trace between asterisk and netborder |
| 09:42.44 | fredericve | I clearly see the udptl flow |
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| 09:43.10 | bulkorok | then check the messages and try to see the problem |
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| 09:52.31 | fredericve | ok, what kind of problem am I looking for? and what is the recommended maxdatagram size? |
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| 11:17.26 | mirela666 | Hello, I'm getting a wierd error in log: Call from 'sipPeer' to extension 'extension' rejected because extension not found. |
| 11:17.53 | mirela666 | [May 30 13:07:56] NOTICE[27488] chan_sip.c: Peer 'sipPeer' is now Reachable. (31ms / 2000ms) |
| 11:18.22 | mirela666 | [May 30 13:07:56] VERBOSE[3675] logger.c: -- Added extension '_extensionX.' priority 1 to from-MS |
| 11:19.11 | mirela666 | error is after reload, |
| 11:19.22 | Greenlight | Your sending the call to "extension" - Does that extension exist? |
| 11:22.15 | mirela666 | yes, but in pattern it's _mondX. is n restricted for N any digit? |
| 11:25.10 | kaldemar | _extensionX. does not match 'extension' in any way. |
| 11:25.16 | Greenlight | Indeed |
| 11:25.28 | leifmadsen | _e[x]te[n]sio[n]. |
| 11:25.29 | kaldemar | even length does not match. |
| 11:25.31 | leifmadsen | that will work |
| 11:25.38 | robl^ | and have you made sure the context is correct? |
| 11:25.39 | leifmadsen | and do what you intend |
| 11:25.43 | leifmadsen | (add the X on the end for the number) |
| 11:25.56 | leifmadsen | lowercase and upper case are not relevant to pattern matches |
| 11:25.59 | Greenlight | Doesn't look like there is a number on the end |
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| 11:26.14 | leifmadsen | <PROTECTED> |
| 11:26.24 | leifmadsen | _e[x]te[n]sio[n]X. |
| 11:26.25 | kaldemar | and the pattern requires two or more after 'extension' |
| 11:26.29 | Greenlight | Call from 'sipPeer' to extension 'extension' rejected because extension not found. |
| 11:26.36 | Greenlight | ^^ Thats error he pasted |
| 11:26.44 | mirela666 | yes xX nN reserved |
| 11:28.39 | mirela666 | I just gave an example, cause of my low knowledge :) extension was _mondX. and dialed mond1234 |
| 11:29.19 | mirela666 | and _mo[1-9]dX. was expected, sorry for my "knowledge" |
| 11:29.23 | Greenlight | So you changed the error message before posting ? |
| 11:29.37 | mirela666 | yes, yes |
| 11:29.41 | Greenlight | Pfft I'm out |
| 11:29.57 | mirela666 | only the name of the peer and extensions |
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| 11:33.33 | fredericve | bulkorok: If I compare the udptl traces between a working setup and the malfunctioning setup, the packets on the malfunctioning one in wireshark all show "malformed packet" |
| 11:35.04 | bulkorok | fredericve: maybe a hardware problem!? |
| 11:35.24 | fredericve | I'm suspecting that too now |
| 11:36.01 | bulkorok | google found: When there are extra octets after the T.38 UDPTL packet you will now see |
| 11:36.03 | bulkorok | [Malformed?] in the Info-column |
| 11:36.35 | bulkorok | but at first I would check hardware... cables at first |
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| 11:46.29 | skrusty | Just in case anyone is interested, the Asterisk.NET project has been re-established on Codeplex (asternet.codeplex.com) and is currently under active development! |
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| 12:08.00 | thom| | bonjour |
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| 12:25.20 | robsonpeixoto | Hi. I'm noob in asterisk but I need to develop a FastAGI/AGI. I'd like to know what's is the more stable API to develop AGI. Thanks |
| 12:26.37 | skrusty | depends on language |
| 12:26.47 | skrusty | what are you wanting to code in? |
| 12:27.26 | [TK]D-Fender | I've never heard of an "unstable" AGI API |
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| 12:28.13 | skrusty | i assume he's refering to the framework being used against FastAGI :) |
| 12:32.30 | [TK]D-Fender | that's just basic TCP... I don't see anything complex or different in there as functionality goes that should bring this into question |
| 12:34.28 | skrusty | not sure i agree, although the AGI protocol may be well established, frameworks for using it (e.g. ASterisk-Java or Asterisk.NET) may not be complete or may be classed as "unstable" |
| 12:34.31 | robsonpeixoto | skrusty, I'm agnostic about language |
| 12:35.20 | skrusty | robsonpeixoto: well i am a little bias, but there aew a few, ASterisk-Java and AsterNET (for .net). There are more, but I am not as well versed with them |
| 12:35.28 | robl^ | Another concern should be considered is the overhead brought with the framework/language runtime. i.e. java is quite a bit heavier (resource-wise) than say something using PHP or perl. |
| 12:35.41 | skrusty | I suggest you have a little look on the voip-info.org wiki, there's information there that may help you decide |
| 12:36.08 | robsonpeixoto | skrusty, Do you recommend the starpy ? |
| 12:36.28 | skrusty | never used it |
| 12:36.36 | [TK]D-Fender | robsonpeixoto: What do you actually want to do? |
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| 12:37.58 | robsonpeixoto | [TK]D-Fender, With data typed by the user, I'll need to make some query on my database e say the result. [sorry my bad english] |
| 12:38.25 | [TK]D-Fender | robsonpeixoto: AGI may not even be required for this... |
| 12:38.35 | [TK]D-Fender | robsonpeixoto: func_odbc <- |
| 12:38.37 | robl^ | robsonpeixoto" look at function odbc |
| 12:39.30 | robsonpeixoto | [TK]D-Fender, using AEL. With AEL can I access REST Webservice ? |
| 12:40.55 | robl^ | for REST, look at func_curl ;-) |
| 12:41.24 | [TK]D-Fender | AEL does nothing special |
| 12:41.30 | robsonpeixoto | A good doc to learn AEL. Are there? |
| 12:41.38 | [TK]D-Fender | it is just an alternative syntax front-end to standard extensions.conf |
| 12:41.47 | [TK]D-Fender | And is one more thing that can break on your system |
| 12:41.54 | skrusty | :) |
| 12:42.38 | kaldemar | robsonpeixoto: AEL does not provide you REST access any more than regular dialplan does. |
| 12:43.51 | robl^ | some of us like to live on the edge, and use neither AEL2 or extensions.conf dialplans in lua ;) |
| 12:47.23 | robsonpeixoto | But the func_curl and func_odbc are from AEL? How can I create scritps with a litter complex logic like: Consult a webservice, use the text to TTS and play the result of TTS without a AGI. [I'm very noob.rsrs] |
| 12:48.23 | [TK]D-Fender | robsonpeixoto: They have nothing to do with AEl |
| 12:48.42 | [TK]D-Fender | robsonpeixoto: You are advised to very quickly forget you ever heard of AEL right now.... |
| 12:48.47 | *** join/#asterisk MrQuist (~rob@83.232.96.217) |
| 12:48.54 | MrQuist | Hey guys, i have a small question |
| 12:49.09 | MrQuist | I'm trying to create a nightmode on 1 extension (450). |
| 12:49.19 | [TK]D-Fender | robsonpeixoto: It does not offer any EXTRA functionality. |
| 12:49.44 | robsonpeixoto | [TK]D-Fender, thanks =D I forgot |
| 12:50.17 | MrQuist | This is my extension list: |
| 12:50.18 | MrQuist | http://pastebin.com/VB5yhmp1 |
| 12:50.27 | MrQuist | if i dial 450, i get hung up instantly |
| 12:50.30 | *** join/#asterisk Draecos (~Draecos@124-168-247-223.dyn.iinet.net.au) |
| 12:51.21 | [TK]D-Fender | MrBecause you passed 1 arguement after the "?" |
| 12:51.39 | [TK]D-Fender | MrQuist: Which means it's looking for a PRIORITY named "nightmode" |
| 12:51.43 | MrQuist | yeah, so? |
| 12:51.51 | MrQuist | oh, is'nt that an extension? |
| 12:51.55 | [TK]D-Fender | MrQuist: NO. |
| 12:51.59 | MrQuist | OK |
| 12:52.09 | MrQuist | I TOOK A LOOK AT THIS EXAMPLE: |
| 12:52.11 | [TK]D-Fender | exten => 450,1,GotoIfTime(9:00-19:33,mon-fri,*,*?nightmode) <---- Priority 1 (step) |
| 12:52.28 | [TK]D-Fender | MrQuist: It's loking for a label on a step in the current extensions |
| 12:52.40 | [TK]D-Fender | MrQuist: "core show application gotoif" |
| 12:52.43 | MrQuist | http://www.voip-info.org/wiki/view/Asterisk+day+night+mode+example |
| 12:53.03 | MrQuist | aaah |
| 12:53.05 | [TK]D-Fender | exten => 4710,2,GotoIfTime(12:00-12:59|*|*|*?night-mode|1) <- notice the |1 on the end? |
| 12:53.08 | [TK]D-Fender | You didn't do that |
| 12:53.10 | MrQuist | Yes i did |
| 12:53.20 | [TK]D-Fender | so your sample did not actually try to copy what that one showed\ |
| 12:53.28 | MrQuist | I kind of combined that first link with this one: http://www.voip-info.org/wiki/view/Asterisk+cmd+GotoIfTime |
| 12:53.33 | [TK]D-Fender | exten => 450,1,GotoIfTime(9:00-19:33,mon-fri,*,*?nightmode) <- no yours didn't |
| 12:53.43 | MrQuist | no, i know, because its not in the 2nd link |
| 12:53.45 | [TK]D-Fender | that is not n"nightmode|1" |
| 12:53.51 | MrQuist | i know... |
| 12:54.33 | [TK]D-Fender | Keeping in mind that "|" is not a valid delimiter in the dialplan since many versions ago |
| 12:54.55 | MrQuist | So the 2nd link i sent, with the "," instead of | delimiters was correct? |
| 12:55.20 | MrQuist | exten => 450,1,GotoIfTime(17:00-8:59,*,*,*?nightmode,1) ; |
| 12:55.23 | [TK]D-Fender | The WIKI link? |
| 12:55.26 | MrQuist | exten => night-mode,1,Answer |
| 12:55.27 | MrQuist | exten => night-mode,2,Wait(2) |
| 12:55.31 | [TK]D-Fender | indeed it should be "," |
| 12:55.38 | MrQuist | i need to add this one; |
| 12:55.39 | MrQuist | exten => 4710,4,Goto(night-mode,1) |
| 12:56.14 | *** join/#asterisk serafie (~erin@nat/digium/x-qrnjaavwprpcpuyi) |
| 12:56.30 | [TK]D-Fender | you should just code this all within the same exten unless you int3end to share taht code with some other exten |
| 12:56.34 | MrQuist | In the wiki, where can i find that "priority step label thing" |
| 12:56.47 | [TK]D-Fender | ? book |
| 12:56.49 | [TK]D-Fender | ^^ |
| 12:56.52 | [TK]D-Fender | ~book |
| 12:56.52 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 12:56.54 | [TK]D-Fender | ^^ |
| 12:56.57 | MrQuist | Yes, i intend to share that |
| 12:57.04 | [TK]D-Fender | Forget the vpoi-info wiki for these basics |
| 12:57.18 | MrQuist | Okay... |
| 12:57.42 | MrQuist | So, there's no way to: "If time is 17:00 - 09:00 dial ext Nightmode" |
| 12:58.03 | [TK]D-Fender | it is a decrepit mess that should used for minor technical bits once you've got the rest of things down and then fail to find what you're looking for in the tarball'd documentations or on the official Asterisk WIKI |
| 12:58.06 | [TK]D-Fender | wiki.asterisk.org |
| 12:58.32 | [TK]D-Fender | [08:57]MrQuistSo, there's no way to: "If time is 17:00 - 09:00 dial ext Nightmode" <- first your use of the term "dial" is inappropriate, and YES, GotoIfTime works |
| 12:58.45 | [TK]D-Fender | You gave it the wrong target |
| 12:59.01 | MrQuist | Okay.. |
| 12:59.33 | *** join/#asterisk serafie1 (~erin@user-24-214-173-250.knology.net) |
| 13:00.11 | MrQuist | I though voip-info would be a good source of information |
| 13:00.15 | MrQuist | plus, its on top in google |
| 13:00.23 | MrQuist | but i'll keep in mind that its shit |
| 13:00.30 | MrQuist | also, go grab yourself a cup of coffee :) |
| 13:01.09 | [TK]D-Fender | It used to be good, and it's been around since forever... |
| 13:01.24 | robl^ | MrQuist: the voip-info wiki has LOTS of information, but its not well maintained in regards to Asterisk. Much of it is out of date and no longer acurage. Asterisk has evolved faster than the wiki |
| 13:01.42 | MrQuist | Hi robl^, yeah, i noticed :P |
| 13:02.13 | [TK]D-Fender | that means lots of links. then again most of it's samples like this are OLD code that needs to be fixed for current versions, or sample contributed by well-meaning schmucks who barely knew what they were doing and are full of little mistakes. |
| 13:03.17 | MrQuist | Okidokie :) thanks |
| 13:03.23 | *** join/#asterisk clh (~clh@38.110.19.123) |
| 13:03.40 | [TK]D-Fender | First, understand your dialplan basics ... the BOOK I linked is good for that. Then read your apps INSTRUCTIONS from * CLI "core show application X", "core show function X", etc |
| 13:04.17 | [TK]D-Fender | And make sure you understand how to pass context,exte,priority(label) to apps that require them. |
| 13:06.55 | MrQuist | Great |
| 13:07.16 | MrQuist | It works |
| 13:08.29 | MrQuist | All i needed to do was, add an argument to GotoIfTime (1) |
| 13:08.41 | MrQuist | Change exten => nightmode,1,Answer() with exten => nightmode,1,NoOp() |
| 13:09.12 | MrQuist | i just haven't worked with goto's and jumps yet, i am sorry |
| 13:09.43 | MrQuist | the reason i came here is because the code seemed to fail and i didn't get any errors in asterisk. |
| 13:09.55 | MrQuist | And im used to code throwing exceptions |
| 13:09.58 | MrQuist | instead of nothing |
| 13:10.45 | [TK]D-Fender | no target to land on => hangup |
| 13:11.21 | MrQuist | and with target you mean? An extension? |
| 13:11.31 | [TK]D-Fender | GotoIf, Goto, etc. |
| 13:11.37 | MrQuist | Aha. |
| 13:11.38 | [TK]D-Fender | Anywhere you specific to go somewhere else |
| 13:11.53 | MrQuist | Thats weird because i didn't 'add' a target |
| 13:12.04 | MrQuist | i changed Answer() to NoOp() in the nightmode extension. |
| 13:12.40 | [TK]D-Fender | Doesn't really matter |
| 13:12.53 | [TK]D-Fender | The Playback on the next priority will Answer() the call anyway |
| 13:12.58 | MrQuist | See the difference: http://pastebin.com/2vYjqv48 |
| 13:13.20 | Katty | i'll play back your next priority call in a minute |
| 13:13.35 | MrQuist | then it was the ,1 in the GotoIfTime |
| 13:13.39 | MrQuist | Where i was missing an argument |
| 13:13.51 | MrQuist | Which is weird because i didn't get any parse errors or something |
| 13:14.07 | kaldemar | MrQuist: you weren't really missing an argument, your argument was of wrong type. |
| 13:14.48 | kaldemar | MrQuist: "nightmode" would mean n(nightmode) in the same extension, i.e. a label. "nightmode,1" would mean extension nightmode, priority 1. |
| 13:14.50 | MrQuist | and.. what wrong type did i passed to it? "NULL" ? |
| 13:15.19 | MrQuist | So, "nightmode" would be extension nightmode, unknown priority => hangup |
| 13:15.27 | MrQuist | correct? |
| 13:15.28 | [TK]D-Fender | MrQuist: The you weren't missing an argument... you simply gave it a valid one ... whose target didn't exist |
| 13:15.34 | kaldemar | not type as in datatype, but label vs. extension and priority. |
| 13:15.47 | MrQuist | very weird |
| 13:15.47 | kaldemar | MrQuist: wrong. |
| 13:15.59 | MrQuist | GotoIfTime(9:00-19:33,mon-fri,*,*?nightmode) => HANGUP |
| 13:16.09 | MrQuist | GotoIfTime(9:00-19:33,mon-fri,*,*?nightmode,1) => Go to extension nightmode |
| 13:16.13 | MrQuist | Why? |
| 13:16.14 | kaldemar | <PROTECTED> |
| 13:16.21 | MrQuist | Ah like that |
| 13:16.25 | kaldemar | MrQuist: n(LABEL) |
| 13:16.36 | MrQuist | first one == GotoIfTime(9:00-19:33,mon-fri,*,n(nightmode),1) |
| 13:16.44 | MrQuist | without the latter -1 |
| 13:16.48 | kaldemar | wrong again. |
| 13:16.52 | MrQuist | damnit |
| 13:17.04 | MrQuist | i'm a php programmer, i don't understand this freaking syntax |
| 13:17.15 | kaldemar | you had it correct by syntax already, i just told you what they mean. |
| 13:17.21 | [TK]D-Fender | MrQuist: http://pastebin.com/UZMnxTDg |
| 13:17.36 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.135) |
| 13:17.53 | MrQuist | yes, i ment that |
| 13:18.12 | MrQuist | and if i add ,1 on the end |
| 13:18.24 | MrQuist | it searches for an extension "nightmode" in stead of a label nightmode |
| 13:18.28 | [TK]D-Fender | MrQuist: context,exten,priority(or label) |
| 13:18.34 | [TK]D-Fender | MrQuist: exten,priority(or label) |
| 13:18.45 | [TK]D-Fender | MrQuist: priority(or label) |
| 13:18.54 | MrQuist | Great |
| 13:18.57 | MrQuist | i understand it now |
| 13:18.59 | [TK]D-Fender | MrQuist: you specify in order of precision |
| 13:19.02 | MrQuist | Thanks a lot |
| 13:19.04 | MrQuist | yes i see |
| 13:19.18 | MrQuist | but its just, i still don't know why there would be a "*?" before "nightmode" |
| 13:19.24 | *** join/#asterisk jacobw (~jacob@unaffiliated/jacobw) |
| 13:19.37 | [TK]D-Fender | Mrin the Gotoif? |
| 13:19.51 | *** join/#asterisk davlefouAMD (~david@41.225.42.27) |
| 13:19.56 | kaldemar | MrQuist: have you taken a look what "core show application GotoIfTime" in asterisk's CLI says? |
| 13:19.58 | [TK]D-Fender | MrQuist: GotoIf(time) |
| 13:20.33 | [TK]D-Fender | MrQuist: "?" separates the TEST on the left from the DESTINATION on the right |
| 13:20.46 | [TK]D-Fender | MrQuist: Consider "?" = "then" |
| 13:21.24 | [TK]D-Fender | MrQuist: and the "*" is not functionally associated with the "?" |
| 13:21.46 | kaldemar | by syntax it's like ternary. even php supports those. |
| 13:22.45 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:22.59 | *** join/#asterisk ctaloi (~ctaloi@50.56.202.179) |
| 13:29.35 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 13:29.50 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:30.07 | MrQuist | $var = ($a == b ? true : false); |
| 13:30.09 | MrQuist | those you mean? |
| 13:30.24 | MrQuist | i get it now :) |
| 13:30.28 | MrQuist | thanks kaldemar and [TK]D-Fender |
| 13:31.10 | Greenlight | Although that would more aptly be written as "$var = $a == b" wouldn't it? |
| 13:31.29 | [TK]D-Fender | Greenlight: SHHH!! |
| 13:31.35 | Greenlight | hides |
| 13:31.48 | [TK]D-Fender | Greenlight: we're debating function, not application ;) |
| 13:32.30 | Greenlight | It's like code-tourettes; I can't help but optimise |
| 13:33.02 | MrQuist | Greenlight, in that case it'd be $var = ($a == b); |
| 13:33.24 | Greenlight | Brackets are optional |
| 13:33.37 | [TK]D-Fender | Greenlight: that's backwards... Tourette's would mean you're adding extra syntax where it isn't required :) |
| 13:33.55 | Greenlight | Well in this case, optimising when it's not required :) |
| 13:35.53 | MrQuist | oh, i always use those brackets |
| 13:35.56 | MrQuist | because of readability |
| 13:36.04 | MrQuist | but i guess you're right :P |
| 13:36.25 | Greenlight | Yea, I agree it does make the intenation clearer |
| 13:36.52 | MrQuist | i'll be spanked hard if i don't comply to the PSR-2 code standard |
| 13:37.25 | Greenlight | So, is that a punishment or reward ? :) |
| 13:37.42 | [TK]D-Fender | YES |
| 13:37.48 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 13:42.21 | *** join/#asterisk MatBoy (~yamakasi@62.58.32.94) |
| 13:42.27 | MrQuist | Hahahah |
| 13:42.41 | MrQuist | So, now that i have the automatic night-mode configured |
| 13:42.56 | MrQuist | How could i make a person be able to avoid the nightmode configuration? |
| 13:42.57 | MatBoy | hi guys, is there a manager URL for asterisknow like: 'http://asteriskmanagerhostname.domain:8088/asterisk' |
| 13:43.25 | MrQuist | By dialing a special number |
| 13:43.27 | MatBoy | I only have the admin URL for the panel |
| 13:43.37 | Greenlight | MrQuist: As in, for a user to be able to manually enable/disable it, or for a caller to avoid it? |
| 13:43.48 | MrQuist | The latter |
| 13:44.07 | MatBoy | port 8088 is also not open on asteriskNow |
| 13:44.11 | Greenlight | MatBoy: AsteriskNow has two choices of GUI; which did you choose? |
| 13:44.28 | Greenlight | Hint; FreePBX is the one you want |
| 13:44.35 | MatBoy | Greenlight: I didn't chose one, it installed FreePBX itself :\ |
| 13:44.45 | MrQuist | yeah we've used freepbx, works ... okay |
| 13:44.56 | Greenlight | MatBoy: Then it's the default port 80 |
| 13:45.12 | MatBoy | Greenlight: but there is no /asterisk folder ? |
| 13:45.13 | Greenlight | MrQuist: Either a special number, "hidden" IVR, or specific DDI |
| 13:45.20 | Greenlight | MatBoy: No |
| 13:45.27 | MatBoy | Greenlight: okay thanks |
| 13:45.31 | MrQuist | Wait, Greenlight, i mean a general enable / disable functionality |
| 13:45.34 | MrQuist | company-wide |
| 13:45.58 | Greenlight | Ahh - we have a similar thing to enable/disable our "unforceen circumstances" messages |
| 13:46.01 | MrQuist | So i can dial e.g. 272* and disable the day/night mode |
| 13:46.07 | Greenlight | A special number, and PIN code |
| 13:46.14 | MrQuist | Exactly, such a thing |
| 13:46.25 | MrQuist | allthough we're not that big, so a pin code won't be neccecary |
| 13:46.31 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 13:46.33 | Greenlight | I recommend the PIN if you users are anything like mine |
| 13:46.41 | MrQuist | We're all programmers |
| 13:46.48 | Greenlight | Oh, 100% need a PIN! :) |
| 13:46.52 | MatBoy | Greenlight: but that URL also works for the API ? because that is what I need |
| 13:47.00 | MrQuist | Is that so? :P haha okay |
| 13:47.01 | Greenlight | MatBoy: What API ? |
| 13:47.17 | MatBoy | Greenlight: wasn't there an API ? |
| 13:47.21 | MatBoy | for asterisk ? |
| 13:47.31 | MrQuist | Asterisk Programming INterface |
| 13:47.36 | Greenlight | MrQuist: If I recall correctly, we use a global variable, and set that via the the spceial number |
| 13:47.38 | MrQuist | yeah, apt-get install ssh |
| 13:47.57 | MrQuist | then in the GotoIfTime, add another condition |
| 13:48.12 | Greenlight | Yes, or before the GotoIfTime |
| 13:48.14 | *** join/#asterisk ChadAragorn (~ChadArago@206.251.40.221) |
| 13:48.16 | MrQuist | Right |
| 13:48.17 | Greenlight | SO you can skip it |
| 13:48.28 | MrQuist | If d_n == true -> check time -> do stuff |
| 13:48.34 | MrQuist | if d_n == false -> continue to call |
| 13:48.38 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:48.40 | Greenlight | Exactly |
| 13:48.44 | MrQuist | Great |
| 13:49.06 | Greenlight | SO, have a label *after* the time check where it would continue, and goto there if "skipping" |
| 13:49.13 | MrQuist | Yes indeed |
| 13:49.29 | Greenlight | There may well be a slicker way to do it that a global variable, but that works for us |
| 13:49.40 | MrQuist | yeah that'll do just fine here |
| 13:49.49 | MrQuist | Thanks :) |
| 13:49.50 | Greenlight | One thing to watch is that I *think* a reload resets it to it's default state |
| 13:49.57 | msaraiva | MatBoy: AMI -> Asterisk Manager Interface. You use that to control asterisk from "outside" |
| 13:50.05 | msaraiva | Usually it's running on port 5038 |
| 13:50.17 | msaraiva | See /etc/asterisk/manager.conf |
| 13:50.38 | Greenlight | I think the latest FreePBX builds have that in a GUI |
| 13:50.38 | MatBoy | msaraiva: ah thanks! |
| 13:50.43 | msaraiva | It's probably what you're looking for. |
| 13:50.44 | Greenlight | It'll overwrite manager.conf iirc |
| 13:50.58 | msaraiva | It does, actually. |
| 13:51.03 | msaraiva | But he can see the current parameters there. |
| 13:51.13 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:51.13 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:51.32 | msaraiva | I think freepbx defaults it to listening only on the loopback interface. |
| 13:52.19 | msaraiva | MatBoy: http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 13:52.20 | Greenlight | When you do change that, *dont* open it up on an external interface |
| 13:52.46 | MatBoy | msaraiva: I'm busy with this kind of "integration" that is why I ask: http://www.tine20.org/wiki/index.php/Admins/Asterisk_integration |
| 13:52.46 | chris_n | has anyone successfully added a Geolocation header to sip invites? |
| 13:53.44 | Greenlight | MatBoy: At a glance, I'm unsure that will play well with FreePBX |
| 13:53.53 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:53.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:54.50 | msaraiva | Yep, probably not. |
| 13:55.10 | msaraiva | If you really want to go that way, forget about a GUI. |
| 13:55.15 | msaraiva | Or write one yourself... |
| 13:55.23 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-jqufhjycpzacowos) |
| 13:55.23 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:55.31 | MatBoy | Greenlight: indeed, so I need a native install of Asterisk I guess.. so just install Ubuntu or CentOS and install asterisk ? |
| 13:56.01 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:56.01 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:56.09 | Greenlight | MatBoy: Yes, although it looks like you'll still need to do some dialplan work to get a "working" system |
| 13:56.51 | MatBoy | Greenlight: yeah indeed, but that should be doable... I think freePBX on top of an asterisk installation will do better than just a ISO download as I installed now |
| 13:57.32 | Greenlight | I would recommend a proper install from source |
| 13:58.01 | Greenlight | Install CentOS6, then compile and install Asterisk 11. Should only take 30 minutes, and there's loads of guides online. |
| 13:58.41 | MatBoy | Greenlight: yeah indeed no problem |
| 13:58.48 | MatBoy | Greenlight: an Ubuntu install will do also I think |
| 13:58.51 | msaraiva | FreePBX won't play well with that software. |
| 13:59.10 | Greenlight | Yup, that's what I thought as well |
| 13:59.33 | Greenlight | MatBoy: Yeah, whatever flavour of Linux is your preference |
| 13:59.49 | msaraiva | For it to work, you're going to need Asterisk with a realtime backend for sip peers. |
| 13:59.56 | MatBoy | msaraiva: it's not about FreePBX... the installer was nice... but I like custom installs more |
| 14:00.00 | Greenlight | I think CentOS is probably most popular amongst Asterisk installs but it's up to you |
| 14:00.30 | msaraiva | MatBoy: It's always better to go custom, especially when we're talking about Asterisk. |
| 14:00.59 | msaraiva | What i'm saying is that you won't be able to use "freepbx on top of an asterisk installation" like you said before. |
| 14:01.09 | msaraiva | If you want to use that software you linked, anyway. |
| 14:01.29 | msaraiva | FreePBX messes with the whole Asterisk configuration, and has it's own database. |
| 14:01.46 | Greenlight | eg. You'll need to write your own dialplan to perform any functions you require (like making a call!) |
| 14:02.22 | MatBoy | msaraiva: indeed, I have seen that also in the past, thanks! |
| 14:02.28 | MatBoy | Greenlight: doable! |
| 14:02.50 | msaraiva | Besides, FreePBX writes sip peers to sip.conf from its own database, it doesn't use realtime backend. |
| 14:03.14 | jacobw | FreePBX is horrific |
| 14:03.55 | msaraiva | I think it's fine for it's purpose, to provide some type "regular pbx" functionality. |
| 14:04.07 | msaraiva | But for everything else... |
| 14:04.24 | Greenlight | Yea, it's a good "introduction" I guess. And for small size PBX's can be fine. It scales *very* poorly though. |
| 14:15.44 | [TK]D-Fender | FreePBX is fine if you're happy with a basical SMB layout for most features. If you want a little extra special on top you can add your own custom stuff on the side. If you want to actually change it's starndard flow, that's where it gets really hard... |
| 14:16.50 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 14:24.15 | [TK]D-Fender | unmonitored open AMI =death sentence |
| 14:26.07 | MrQuist | Sweet, Greenlight, it works :) |
| 14:26.12 | MrQuist | [globals] |
| 14:26.12 | MrQuist | DAYNIGHTENABLED=1 |
| 14:26.28 | MrQuist | exten => 460,1,GotoIf($[${GLOBAL(DAYNIGHTENABLED)} = 1]?disable:enable) |
| 14:26.28 | MrQuist | same => n(enable),Set(GLOBAL(DAYNIGHTENABLED)=1) |
| 14:26.28 | MrQuist | same => n(disable),Set(GLOBAL(DAYNIGHTENABLED)=0) |
| 14:26.46 | MrQuist | and ofcourse the exten => 450,1,GotoIf($[${GLOBAL(DAYNIGHTENABLED)} = 1]?timecheck:call) |
| 14:27.13 | [TK]D-Fender | MrQuist: bad code |
| 14:27.28 | MrQuist | Oh jees i'm sorry |
| 14:27.28 | [TK]D-Fender | MrQuist: your 2 sets are back to back |
| 14:27.35 | [TK]D-Fender | and PASTEBIN <-------- |
| 14:27.35 | MrQuist | i know |
| 14:27.50 | MrQuist | i'd do -> Set(VAR = !VAR); |
| 14:28.04 | MrQuist | but i need an audial response |
| 14:28.12 | Greenlight | Yea, as [TK]D-Fender picked up on, it'll always set disabled |
| 14:28.17 | Greenlight | If it flows like you paste |
| 14:28.23 | MrQuist | Wat? it doesn't |
| 14:28.30 | MrQuist | there's a hangup and playback under the Set |
| 14:28.34 | Greenlight | Ahh |
| 14:28.35 | Greenlight | :) |
| 14:28.40 | Greenlight | Hid that from us you did :) |
| 14:28.41 | [TK]D-Fender | MrQuist: not that you showed right above |
| 14:28.44 | MrQuist | Set -> Playback -> Hangup -> Set |
| 14:28.46 | Greenlight | Glad it's working |
| 14:28.51 | MrQuist | no i didn't want to paste the whole code |
| 14:28.56 | MrQuist | it was on the edge of pastebinning |
| 14:28.57 | Greenlight | Also, there's some nice audio files bundled that you can use |
| 14:28.59 | [TK]D-Fender | Greenlight: Talks does Yoda funny hhhmmmMMMM!?!?!??!?!? |
| 14:29.25 | Greenlight | :) |
| 14:29.31 | MrQuist | Greenlight, i've yet to find them - i used digits/1 and digits/0 :P |
| 14:29.44 | MrQuist | but the basic functionality is there |
| 14:29.59 | MrQuist | Its not that hard |
| 14:30.02 | Greenlight | If you're anything like me you'll loose half your day listening to all the comical audio files included |
| 14:30.08 | MrQuist | it just reads real shitty |
| 14:30.11 | MrQuist | hahahah yeah i did |
| 14:30.15 | MrQuist | monkeys.. Wtf is that? |
| 14:30.19 | MrQuist | Oooooh.. Cool! |
| 14:30.33 | MrQuist | combine with -> something went terribly wrong -> monkeys -> beeps -> hangup |
| 14:30.46 | MrQuist | We currently have 692 linked to batman.mp3 |
| 14:30.51 | Greenlight | You found the zombies one yet? |
| 14:30.55 | [TK]D-Fender | I triggered a Page to my marketing dept with 4 phones on auto-answer and sent monkeys their way.... |
| 14:30.55 | MrQuist | nope |
| 14:31.15 | Greenlight | [TK]D-Fender: Ammused am I |
| 14:31.18 | [TK]D-Fender | The entire room was freaking out |
| 14:31.50 | MrQuist | lol |
| 14:32.26 | igcewieling | I hate hosted clients |
| 14:32.33 | igcewieling | hate them hate them hate them! |
| 14:33.05 | Greenlight | I believe there's an "activated" and "cancelled" audio files (unless someone at my side chopped those from existing files) |
| 14:33.34 | Greenlight | That's what we use for the enabled/disabled messages |
| 14:33.53 | Greenlight | igcewieling: Why, what's happened? |
| 14:34.39 | *** join/#asterisk jm|laptop (~jm|laptop@null.jamiem.com) |
| 14:34.46 | jm|laptop | hello :) |
| 14:35.11 | igcewieling | Greenlight: It would take a month to describe all the problems with a hosted service |
| 14:35.43 | igcewieling | todays issue us "phones are randomly rebooting" at a couple of clients. I can't tell if they just being drama queens or if something else is going on. |
| 14:35.52 | jm|laptop | I'm running Asterisk 1.8.13.1~dfsg-3. Why might be making it ignore the context for a peer and keep trying default? |
| 14:36.10 | jm|laptop | I have type=peer context=the-context |
| 14:36.17 | Greenlight | igcewieling: I know the "drama queen" / "actual problem" dilemma all too well |
| 14:36.21 | igcewieling | jm|laptop: the incoming call did not match any peers |
| 14:36.52 | jm|laptop | igcewieling: the peer= has a FQDN is it that the incoming IP did not match the rDNS or similar? |
| 14:37.12 | MrQuist | Hey Greenlight, i wonder - i currently have that day/time switch hooked up to extension 460 |
| 14:37.15 | jm|laptop | I am led to believe the peer authenticates, too |
| 14:37.22 | MrQuist | but, can i make it something like *372# ? |
| 14:37.24 | igcewieling | Greenlight: the hosted customers are dumber than a bag of rocks. If they were not, they would have chosen the non-hosted service. Since they are dumber than a bag of rocks, trying to get them to help with troubleshooting is a waste of time. |
| 14:37.25 | [TK]D-Fender | jm|laptop: show us the actual peer, and the actual call |
| 14:37.27 | [TK]D-Fender | ~pb |
| 14:37.28 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:37.29 | [TK]D-Fender | ^^^ |
| 14:37.47 | Greenlight | MrQuist: Sure, should be able to |
| 14:37.52 | [TK]D-Fender | jm|laptopI am led to believe the peer authenticates, too <- telephony is not "faith-based" |
| 14:38.07 | jm|laptop | no indeed |
| 14:38.44 | jm|laptop | k |
| 14:38.51 | igcewieling | JESUS IS MY ITSP! |
| 14:38.57 | Greenlight | igcewieling: I feel your pain |
| 14:39.59 | *** join/#asterisk drenda (~Utente@81.174.38.26) |
| 14:40.21 | drenda | Hi guys |
| 14:40.55 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 14:43.14 | drenda | I've installed Asterisk 11.4.0 but I've a problem saving voicemail in db with ODBC. I made it many times with previous version of Asterisk but not I've this errror |
| 14:43.31 | drenda | [2013-05-30 16:43:19] WARNING[8578][C-00000005]: app_voicemail.c:4073 insert_data_cb: SQL Direct Execute failed! |
| 14:43.31 | drenda | [2013-05-30 16:43:19] WARNING[8578][C-00000005]: res_odbc.c:608 ast_odbc_direct_execute: SQL Execute error! Verifying connection to odbc_asterisk [asterisk_mysql]... |
| 14:43.31 | drenda | [2013-05-30 16:43:19] WARNING[8578][C-00000005]: app_voicemail.c:4073 insert_data_cb: SQL Direct Execute failed! |
| 14:43.31 | drenda | [2013-05-30 16:43:19] WARNING[8578][C-00000005]: app_voicemail.c:4189 store_file: SQL Execute error! |
| 14:43.37 | Greenlight | ~pb |
| 14:43.37 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:44.16 | drenda | also if the odbc is correctly configured and it works root@phonex:~# isql asterisk_mysql |
| 14:44.17 | drenda | +---------------------------------------+ |
| 14:44.17 | drenda | | Connected! | |
| 14:44.50 | Greenlight | drenda: Also, if you enable debug mode I think you'll see the actual SQL being sent, and can check what's wrong |
| 14:45.12 | Greenlight | Sounds like mysql's moaning about the syntax or some other constraint |
| 14:45.40 | drenda | Greenlight, i've enabled debug mode but there is no other logs a part of that I paste |
| 14:45.40 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 14:45.47 | MrQuist | Greenlight, could i ask you, what would be an easy way to check a simple PIN code? I read something about Read() but i can't find a simple example |
| 14:46.17 | MrQuist | dial number -> *Enter your pin now* -> incorrect -> hangup |
| 14:46.31 | Greenlight | MrMeek: http://pastebin.com/JYyUfRAN <-- That's how it's done on the server I happened to have open |
| 14:46.55 | drenda | Greenlight, so I enable mysql log and I can see the query that seems correct: 43 Query INSERT INTO MessaggioVocale (dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailboxuser,mailboxcontext,flag,msg_id) VALUES ('/var/spool/asterisk/voice |
| 14:46.55 | drenda | mail/default/40/INBOX','0',...... |
| 14:47.24 | Greenlight | Run the query manually, and observe for errors. It's possible overflow or other type constraints |
| 14:48.11 | [TK]D-Fender | MrQuist: "core show application authenticate" |
| 14:48.13 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 14:48.25 | [TK]D-Fender | MrOr just "read" & check the result.... |
| 14:48.31 | [TK]D-Fender | MrQuist: Or just "read" & check the result.... |
| 14:48.38 | MrQuist | doing that now |
| 14:48.42 | Greenlight | Yea, that's what I do in the example I gave him |
| 14:49.57 | Greenlight | It's actually a macro as we have different diverts, so it's called with say: exten => 9903,1,Macro(divert,8346,GLOBAL(BC_divert),1) |
| 14:51.23 | Greenlight | The arguemnts being <PIN> <GLOBAL VAR NAME> <ENABLE/DISABLE> |
| 14:55.23 | *** join/#asterisk navaismo (~navaismo@189.241.9.57) |
| 14:55.42 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 14:58.33 | MrQuist | Sweet! |
| 14:58.45 | MrQuist | i've done it already by the way |
| 14:58.50 | MrQuist | probably ugly as hell |
| 14:59.04 | Greenlight | Nice :) |
| 14:59.37 | MrQuist | http://pastebin.com/LZZpVQRd |
| 14:59.43 | MrQuist | Read(number) |
| 14:59.48 | MrQuist | gotoIf number == 2233 |
| 14:59.49 | MrQuist | etc. |
| 14:59.53 | MrQuist | But it works :P |
| 15:01.00 | igcewieling | you need to enable query logging on your mysql server to see what queries Asterisk is making |
| 15:01.13 | *** join/#asterisk kresp0 (~kresp0@35.Red-83-39-25.dynamicIP.rima-tde.net) |
| 15:01.15 | MrQuist | remember to turn it off again! |
| 15:01.19 | MrQuist | i made that mistake once |
| 15:01.33 | igcewieling | This is one of the "stupid things" Asterisk does. |
| 15:02.29 | *** join/#asterisk xnice (~xsmile@37.231.122.14) |
| 15:02.54 | [TK]D-Fender | MrQuist: Couple of bugs in there |
| 15:03.17 | drenda | Greenlight, solved: was missing the column msg_id that was added in last version of asterisk. Thanks very much for support! |
| 15:03.26 | xnice | hello...i simply want to use my landline in my mobile...when someone call my landline it will forwarded to my mobile and viseversa |
| 15:03.53 | xnice | i got spa3000 and broadband is that all i need and then just configure it ? |
| 15:04.03 | [TK]D-Fender | xnice: Sure |
| 15:04.04 | Greenlight | drenda: No problems - glad it's sorted! |
| 15:04.32 | drenda | Greenlight, thanks :-) |
| 15:04.50 | xnice | <[TK]D-Fender> what more i need ? server or just free sip services out there ? |
| 15:05.06 | [TK]D-Fender | xnice: Could probably add the auth right in the SPA. |
| 15:05.24 | igcewieling | xnice: this isn't actually a general VoIP/SIP channel |
| 15:06.40 | xnice | igcewieling sorry is there somewhere i can ask these questions ? |
| 15:06.44 | jmetro | hm |
| 15:06.55 | jmetro | my code is not passing arguments |
| 15:07.07 | xnice | <[TK]D-Fender> to search for what i need what should i type in google ? |
| 15:07.10 | igcewieling | xnice: I don't know. Google? |
| 15:07.25 | jmetro | i have a dial local/outboundcontext,20,M(announcecontext,arg1) |
| 15:08.33 | drenda | bye guys |
| 15:08.51 | xnice | when i google i got a lot of things mixed i dont know which one does fit my needs exactly...pstn/fxo some terms there for voip geeks i am newbie and just wants to do use my landline in my mobile and viseversa thats all |
| 15:08.55 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:09.03 | jmetro | Verbose("Local/MyCell@dialSecondary-announce-0000001a;2", "EXTENSION IS MyCell") in new stack |
| 15:09.09 | jmetro | but then later on in the announce context |
| 15:09.32 | jmetro | Verbose("myprovider", "EXTENSION IS ") in new stack |
| 15:09.41 | *** join/#asterisk zoid_ (~awainer@181.29.125.3) |
| 15:10.02 | Greenlight | jmetro: Try adding "/n" to the end of your dialstring |
| 15:10.13 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 15:10.13 | Greenlight | (To prevent optimisation of the local channel) |
| 15:10.22 | Greenlight | Just a guess though |
| 15:10.31 | xnice | is there a freelancer can talk to me private i am willing to pay to help me in this |
| 15:11.26 | jmetro | Greenlight where? |
| 15:11.59 | jmetro | exten => _X.,2,Dial(Local/${EXTEN}@mycompany,20,TtkKgM(mycompany-announceIncoming,${EXTEN})) |
| 15:12.02 | Greenlight | [04:07pm] <jmetro> i have a dial local/outboundcontext,20,M(announcecontext,arg1) <-- so it becomes Local/outboundcontext/n |
| 15:12.29 | Greenlight | Wait that's not right |
| 15:13.15 | Greenlight | Ahh there we go, Local/${EXTEN}@mycompany -> Local/${EXTEN}@mycompany/n |
| 15:13.45 | navaismo | If i create a global var from dialplan, this var is accesible for another channel and affect the value of the previous channel created? |
| 15:13.55 | Greenlight | navaismo: Yes |
| 15:14.14 | Greenlight | It's global and shared |
| 15:14.20 | jmetro | Hah! |
| 15:14.21 | jmetro | that worked. |
| 15:14.26 | Greenlight | Good :) |
| 15:14.28 | jmetro | awesome fix. ++++1 |
| 15:14.43 | navaismo | so its safe to create a Global var_${UNIQUEID} to only affect the new created channel? |
| 15:15.56 | Greenlight | navaismo: Not actually certain if you *need* to declare them in [globals] in extensions.conf |
| 15:16.06 | Greenlight | I always do, but not sure if it's needed |
| 15:16.13 | Greenlight | If you don't then yes, you can do that |
| 15:16.22 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 15:16.27 | navaismo | oki thanks |
| 15:21.15 | jmetro | Greenlight: you had one awesome answer, now tell me you have another.. this one is harder |
| 15:21.22 | Greenlight | heh |
| 15:21.37 | Greenlight | 42? |
| 15:22.19 | [TK]D-Fender | [11:07]xnice<[TK]D-Fender> to search for what i need what should i type in google ? <- nothing... read the MANUAL for it |
| 15:22.39 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 15:24.49 | jmetro | http://pastebin.com/N9sAPerm |
| 15:25.02 | jmetro | i need that first context to resume after the dialsecondary-announce is done |
| 15:25.43 | *** part/#asterisk jm|laptop (~jm|laptop@null.jamiem.com) |
| 15:27.24 | gorkish | what is an inexpensive low effort IVR provider that i can fairly simply integrate with asterisk that has TTS and ASR and stuff all built in for only a couple of simultaneous instances? Im thinking voxeo's hosted offering or something similar |
| 15:28.50 | Greenlight | jmetro: Woudlnt a "reuturn" at end of secondary do it? |
| 15:28.54 | Greenlight | *wouldn't |
| 15:29.00 | Greenlight | *"return" |
| 15:29.10 | jmetro | theres an application that does that? |
| 15:29.24 | Greenlight | Return |
| 15:29.40 | Greenlight | Since it's being called as a Sub, right? |
| 15:29.48 | jmetro | hm..its being Dialed. |
| 15:29.51 | jmetro | i could gosub it. |
| 15:29.54 | jmetro | or goto it |
| 15:30.10 | Greenlight | Oh, it's being dialled |
| 15:30.17 | jmetro | i could certainly switch it to a gosub |
| 15:30.22 | Greenlight | There's an arguemnt for dial which returns to diaplan afterwards |
| 15:30.29 | Greenlight | You could use that |
| 15:30.33 | jmetro | "g" but it returns to current context |
| 15:30.40 | jmetro | and doesnt work apparently [just tried it] |
| 15:30.55 | Greenlight | Use that in conjunction with Return |
| 15:31.09 | jmetro | Gosub + return sounds like the proper method |
| 15:31.21 | leifmadsen | you're using macro |
| 15:31.31 | Greenlight | Oh, yea never noticed that |
| 15:31.34 | leifmadsen | MacroExit() |
| 15:31.39 | leifmadsen | oh you have gosubs in macros? |
| 15:31.39 | leifmadsen | that's weird... but ok |
| 15:31.50 | leifmadsen | GoSub() + Return() is how you return from a subroutine, ya |
| 15:32.01 | jmetro | Im still unfamiliar with how asterisk clasifies things but i have a lot of coding history |
| 15:32.07 | jmetro | and gosub just kinda..happens |
| 15:32.18 | Greenlight | I woulod change those Dial's to GoSubs and use Return |
| 15:33.00 | *** join/#asterisk Tarso (~Tarso@189.61.52.46) |
| 15:34.49 | *** join/#asterisk feeshon (~gaston@ool-45787011.dyn.optonline.net) |
| 15:35.03 | jmetro | hm. Return happens, but it does the same thing Dial did - which is enter something into the CDR and end the call |
| 15:35.41 | Greenlight | Lets see the CLI from a call |
| 15:35.59 | jmetro | relevent portion or.. giant flood or text? |
| 15:36.19 | Qwell | I'm more interested in the dialplan. |
| 15:36.25 | Qwell | I'm betting leifmadsen would be too |
| 15:36.32 | Greenlight | The call in question (or is the system busy with other calls |
| 15:36.35 | Qwell | CLI output can be inferred from that. :D |
| 15:36.51 | Greenlight | He already pastebinned it: http://pastebin.com/N9sAPerm |
| 15:37.01 | Qwell | That's old though. |
| 15:37.18 | Greenlight | Ahh with the Dial -> GoSub changes? |
| 15:37.26 | jmetro | the only change is a moved ; |
| 15:37.33 | jmetro | the gosub was actually above the dials, but commented out |
| 15:37.44 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 15:37.54 | Greenlight | And a "Return" added ? |
| 15:38.09 | jmetro | correct |
| 15:39.27 | Qwell | Which side hung up? |
| 15:40.17 | Qwell | Also, you're not showing the mycompany-announceIncoming context. |
| 15:40.32 | Qwell | Sorry, no, the mycompany context. |
| 15:40.42 | jmetro | hm |
| 15:40.44 | *** join/#asterisk [404] (~wir3@12.179.117.114) |
| 15:40.59 | jmetro | im thinking its a logic issue of just how things are being passed down, but hold on |
| 15:42.22 | Qwell | I suspect that Local channels + g option doesn't work so fantastically. |
| 15:42.26 | jmetro | http://pastebin.com/DNECQyX4 that is the CLI |
| 15:42.29 | [404] | Hello everyone, I am trying to figure out the best solution to have load balancing from incoming calls to multiple asterisk nodes containing a voicemail. I am currently handling outgoing calls using kamailio with weighted balancing. Can this be done with kamailio? |
| 15:43.17 | Qwell | [2013-05-30 10:34:00] WARNING[11127][C-0000006a]: pbx.c:4621 pbx_extension_helper: No application '' for extension (macro-myCompany-FindFollow-announce, s, 11) |
| 15:43.20 | Qwell | O.o |
| 15:43.25 | jmetro | yeah im not sure how its getting that |
| 15:43.40 | Qwell | <PROTECTED> |
| 15:43.59 | jmetro | yes, empty for now |
| 15:44.02 | jmetro | WIP |
| 15:44.08 | jmetro | oh god theres a comma |
| 15:44.22 | Greenlight | hehe |
| 15:44.59 | Greenlight | And on VM below |
| 15:45.13 | jmetro | yeah i saw that too |
| 15:45.14 | Qwell | That'll be $299.94 |
| 15:45.16 | igcewieling | clarification of intent, syntax does |
| 15:45.26 | Qwell | That's Qwell, at Bank of Cayman. |
| 15:45.28 | Greenlight | Wow a $0.06 discount |
| 15:45.46 | zoid_ | Hi, is there any way to see the effective setting for max number of file descriptors from asterisk? I'm running out but lsof shows less than the number I set up with ulimit |
| 15:46.00 | igcewieling | jmetro's issues seem to be fatigue related more than anything else. |
| 15:46.08 | Qwell | zoid_: How did you set it? |
| 15:46.08 | Greenlight | Run ulimit -a as the user asterisk is running as |
| 15:46.21 | zoid_ | Qwell: on the init script |
| 15:46.23 | Greenlight | And set it in /etc/security/limits.conf |
| 15:46.29 | Qwell | How, specifically? |
| 15:46.35 | jmetro | igcewieling: heh.. i left work at 10 PM last night and arrived at work 7AM today and i live an hour away. |
| 15:46.50 | Greenlight | You need more caffine! :) |
| 15:47.05 | jmetro | Caffeine makes me sleepy unless its laced with guarana. |
| 15:47.42 | jmetro | and possibly taurine. |
| 15:47.50 | Greenlight | Red bull then |
| 15:47.55 | jmetro | Bawls usually |
| 15:48.03 | zoid_ | Qwell: setting the variable MAXFILES, that later is used as parameter in ulimit -n (it's ubuntu's script) |
| 15:48.05 | Qwell | I can't find Bawls here. ;( |
| 15:48.19 | zoid_ | Greenlight: doing that shows the expected number (64k) |
| 15:48.19 | jmetro | Thinkgeek has it by the caseful. I ordered it when i lived in no-bawls land. |
| 15:48.22 | Qwell | zoid_: And what makes you think it's hitting a limit? |
| 15:48.31 | Qwell | jmetro: shipping is crazy expensive though |
| 15:48.41 | zoid_ | I get this in the console: res_agi.c:1626 launch_script: Unable to create toast pipe: Too many open files |
| 15:48.43 | Greenlight | zoid_: Which user is asterisk running as ? |
| 15:48.49 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.134) |
| 15:48.53 | zoid_ | and han_sip.c:7041 sip_new: Unable to allocate AST channel structure for SIP channel |
| 15:48.57 | zoid_ | Greenlight: "asterisk" |
| 15:49.11 | Greenlight | And you're 100% sure you did ulimit -a *as asterisk* |
| 15:49.12 | Greenlight | Not as root |
| 15:49.53 | zoid_ | yes, this is what I did: |
| 15:49.57 | zoid_ | root@voicetest1:/etc/asterisk# su asterisk -s /bin/bash |
| 15:49.59 | zoid_ | asterisk@voicetest1:/etc/asterisk$ ulimit -n |
| 15:49.59 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.171) |
| 15:50.00 | zoid_ | 65535 |
| 15:50.24 | Qwell | You didn't su - :( |
| 15:50.30 | Greenlight | +1 |
| 15:50.33 | zoid_ | oops |
| 15:51.04 | Greenlight | I suspect you need to increase the hard limit |
| 15:51.23 | zoid_ | doing su - shows the same number |
| 15:51.30 | Greenlight | Hmm... anyone familiar with the AMI Bridge code, or the Bridge method in general? |
| 15:51.44 | zoid_ | mmm both my soft and hard limits are the same |
| 15:52.02 | Greenlight | zoid_: I would try setting it in limits.conf for the asterisk user |
| 15:52.19 | Greenlight | Unless you've something funky going on and eating descriptors |
| 15:52.27 | zoid_ | Greenlight: I did that |
| 15:52.35 | zoid_ | Greenlight: I'm suspecting of my AGI script |
| 15:52.39 | Greenlight | rebooted? |
| 15:52.45 | zoid_ | no |
| 15:53.01 | _Corey_ | That's spooky... I just had a transcoding server get flaky and die because of a ulimit issue about 15 minutes ago. |
| 15:53.04 | Greenlight | Not sure when it "kicks in" after a change to limits.conf |
| 15:53.05 | zoid_ | should I? |
| 15:53.14 | Greenlight | Hopefuilly someone else will know of hand |
| 15:53.27 | _Corey_ | slowly backs away from keyboard... |
| 15:53.30 | Greenlight | lol |
| 15:53.31 | zoid_ | Greenlight: I'll reboot, just to be sure, It's a test server anyway |
| 15:53.41 | zoid_ | did nothing |
| 15:53.52 | Greenlight | Hmm... anyone familiar with the AMI Bridge code, or Bridging functions in general ? |
| 15:54.19 | [404] | has anyone ever messed with registering your sip provider to a kamailio load balancing for voicemails? |
| 15:55.23 | Greenlight | After an AMI bridge, it's dumping one of the channels back into the dialplan, and I want to know how exactly it decides where to dump it do |
| 15:56.07 | zoid_ | Greenlight: rebooting didn't help |
| 15:56.24 | Greenlight | zoid_: Then something else funky is going on |
| 15:56.30 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 15:57.17 | Greenlight | Are you actually using that many file descriptors? It's not recursivelly calling AGI scripts or something is it ? |
| 15:57.18 | zoid_ | that's odd. I as expecting it to break, I'm benchmarking, but not this way |
| 15:57.53 | Greenlight | When you say, you're benchmarking... you're creating lots and lots of calls ? |
| 15:58.02 | zoid_ | Greenlight: I'm running 5 calls per second benchmark, up to 120 concurrent calls, every one of them launches an AGI script |
| 15:58.37 | Greenlight | It's quite possible that you're using all the fd's up |
| 15:58.58 | Greenlight | If they're not gettin released in a timely manner |
| 15:59.05 | zoid_ | Do you mean something like system-wide? |
| 15:59.10 | Greenlight | Or the AGI script is doing something weird |
| 15:59.44 | Greenlight | 65k does seem a *lot* to consume mind |
| 15:59.51 | zoid_ | the AGI script does weird stuff, but once it finishes, if it left fd's open the OS should release them, am I right? |
| 16:00.09 | Greenlight | It should yes |
| 16:00.09 | Katty | I BLAME CANADA. |
| 16:00.11 | zoid_ | I can pase on pastebin the output of lsof, if it's useful |
| 16:00.25 | Katty | also. |
| 16:00.27 | Greenlight | Yea, what;'s lsof | wc -l say |
| 16:00.28 | Katty | my dear canadian friends. |
| 16:00.37 | Katty | who has a recipe for poutine? |
| 16:00.42 | Katty | surely one of you do. |
| 16:01.08 | zoid_ | Greenlight: 6053 |
| 16:01.15 | Greenlight | Hmmm |
| 16:01.18 | zoid_ | it fluctuates, up to around 8k |
| 16:01.20 | Qwell | Katty: fries, cheese curds, poutine gravy |
| 16:01.52 | Katty | yes, but i need a recipe for poutine gravy. |
| 16:02.07 | Katty | i have taters for friends, and cottage cheese (large curd) since i couldn't find plain ole cheese curds. |
| 16:02.07 | Greenlight | hmm |
| 16:02.08 | zoid_ | ok, It started to yell something ellse: res_timing_timerfd.c:180 timerfd_timer_ack: Call to timerfd_gettime() error: Invalid argument |
| 16:02.14 | Katty | for...fries, i mean. |
| 16:02.20 | Greenlight | zoid_: Do you have an init script set to set the fd's to another value |
| 16:02.21 | Katty | tho i'd be ok if you came of for dinner Qwell |
| 16:02.51 | Greenlight | I seem to recall having a script that was setting them to around the 6k mark |
| 16:02.52 | zoid_ | I don't think so |
| 16:03.09 | Greenlight | You don't see anything like this: |
| 16:03.09 | Greenlight | [root@asterisk ~]# asterisk -r |
| 16:03.09 | Greenlight | Setting max files open to 6506 |
| 16:03.09 | zoid_ | but I'll check, maybe try to start asterisk manually |
| 16:03.57 | Greenlight | There's a setting in asterisk.conf |
| 16:03.59 | Greenlight | "maxfiles" |
| 16:04.01 | Greenlight | Is that set ? |
| 16:04.06 | Katty | speaking of asterisk. |
| 16:04.15 | Katty | does asterisk do okay over a vpn? |
| 16:04.22 | Katty | provided speeds aren't complete crap, and such. |
| 16:04.25 | zoid_ | Greenlight: no and no |
| 16:04.28 | navaismo | Greenlight, is there a way to destroy a Global var? |
| 16:04.40 | Greenlight | C4 will do the trick |
| 16:04.43 | Greenlight | Boom |
| 16:05.12 | navaismo | LOL |
| 16:05.17 | navaismo | so i guess nope |
| 16:05.18 | Greenlight | Erm, aside from that, can't you set it to blank ? |
| 16:05.47 | Greenlight | I think that'll remove it |
| 16:05.48 | navaismo | yes i cant set it to blank but it persist if i do a dialplan show globals the var exist in there |
| 16:07.03 | navaismo | i don't hundreds of blank global variables, i don't known if that affect the performance in the future, so I guess its time to use the AstDb |
| 16:07.11 | Greenlight | In that case I'm not sure |
| 16:07.42 | robl^ | Katty: I've used several VoIP solutions (including Asterisk) over VPN. It works well, as long as there isn't too much latency. |
| 16:07.47 | zoid_ | Greenlight: I seems like setting the variable in asterisk.conf worked, thank you! |
| 16:07.52 | Katty | robl^: cheers. |
| 16:08.09 | Greenlight | zoid_: Ahh that's good.Maybe there was a default there :) |
| 16:08.26 | Greenlight | Or maybe that's was overwriting whatever else was setting it .. |
| 16:08.33 | Greenlight | Either way - it's working |
| 16:08.56 | *** join/#asterisk jkister (~chatzilla@67.200.119.94) |
| 16:09.37 | Greenlight | Hmm back to working out why asterisk dumps channels to "somewhere" in dialplan after a bridge |
| 16:13.17 | Greenlight | Seems it's dumpinmg channels into "s,2" of the context associated with the trunk I dialled out on |
| 16:13.46 | Greenlight | Guess as long as that's consistent and I don't need to use "s,2" for anythign else, I can catch it... |
| 16:15.27 | *** join/#asterisk Rumbles (~Rumbles@host81-149-239-223.in-addr.btopenworld.com) |
| 16:20.10 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
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| 16:56.19 | *** join/#asterisk ruben231 (~OpenDial@112.198.79.101) |
| 16:57.38 | ruben231 | hi guys how do i setup on asterisk 1.8 to used conf bridge rather than DAHDI.Thanks |
| 16:58.11 | WIMPy | Use it |
| 16:58.16 | Greenlight | Ensure the module is installed, and use it. |
| 16:58.17 | Qwell | ruben231: confbridge doesn't exist in 1.8, so... |
| 16:58.31 | WIMPy | Oh, right. |
| 16:58.38 | WIMPy | is not so good at history. |
| 16:58.39 | Greenlight | Oh, is that still "old" ConfBridge? |
| 16:59.09 | Greenlight | I thought the new one was added in 1.8, guess I was wrong |
| 16:59.32 | Qwell | It's not useful in 1.8. |
| 16:59.54 | Greenlight | Yea, that's the horrible old version, it as 10 where the proper one appeared. |
| 17:00.03 | Greenlight | ruben231: Upgrade to Asterisk 10 an option ? |
| 17:00.07 | Greenlight | Or 11 |
| 17:00.10 | Qwell | 10 is dead. |
| 17:00.20 | Greenlight | Already? |
| 17:00.28 | WIMPy | The old one was usable, but indeed rather restricted. |
| 17:00.38 | *** join/#asterisk raden (~Jon@24-240-51-238.dhcp.stpt.wi.charter.com) |
| 17:00.47 | raden | good afternoon Miss Katty |
| 17:00.49 | [TK]D-Fender | 10 has been dead for several months now |
| 17:00.54 | ruben231 | Greenlight: asterisk 1.8 is being used only for the app im using.. |
| 17:01.06 | raden | [TK]D-Fender, why is it dead ? |
| 17:01.10 | Greenlight | Seems like 10 was released only yesterday |
| 17:01.19 | raden | im still on 1.6 |
| 17:01.21 | Greenlight | Doesn't time fly eh |
| 17:01.35 | [TK]D-Fender | raden: Because it was EOL'd |
| 17:01.44 | raden | Whats current ? |
| 17:01.44 | [TK]D-Fender | raden: No more fixes. It's dead. |
| 17:01.47 | [TK]D-Fender | 11 |
| 17:01.49 | Greenlight | ruben231: If you can't upgrade, then whats the issue with meetme ? |
| 17:01.50 | raden | geesh |
| 17:01.56 | [TK]D-Fender | and 8 will be supported for a while as well |
| 17:02.24 | Greenlight | You know that MeetMe doens't require a hardware card? |
| 17:03.26 | Qwell | ~asterisk versions |
| 17:03.27 | infobot | Asterisk versions and their support levels are documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 17:03.53 | Greenlight | still remembers when 1.6 was "cutting edge" |
| 17:04.38 | igcewieling | still remembers when 0.65 was cutting edge |
| 17:04.50 | Greenlight | :) |
| 17:05.38 | WIMPy | hardly remembers using 11 |
| 17:05.50 | Greenlight | I really wish I'd discovered Asterisk a few years earlier and not wasted so much time installing and maintaining all that Avaya crap |
| 17:06.15 | ruben231 | <PROTECTED> |
| 17:06.56 | Greenlight | ruben231: There is a version of ConfBrdige in 1.8, but to be hoenst I think MeetMe is better than that version. |
| 17:07.20 | [TK]D-Fender | yup |
| 17:08.02 | WIMPy | For certain definitions of better. It certainly lacked a lot of features. |
| 17:08.11 | Greenlight | The new ConfBridge is particularly good, but you don't have that in 1.8 |
| 17:08.32 | ruben231 | ok got it..ill keep it in mind |
| 17:08.32 | Greenlight | WIMPy: I seem to recall having a number of issues with the old ConfBridge |
| 17:09.03 | Greenlight | Anyways - time to leave the office - laters! |
| 17:09.16 | WIMPy | I probably remember issues with most things I tried. |
| 17:10.11 | zamba | have you guys seen these: chan_sip.c:22546 in handle_request_invite: Sending fake auth rejection for device 'or''=''or''='<sip:'or''=''or''='@<local ip>? |
| 17:12.17 | [TK]D-Fender | zamba: is that actual optupt, or pseudo'd for us? |
| 17:13.01 | [TK]D-Fender | zamba: because fake rejections are perfectly NORMAL and done for security reasons. |
| 17:13.25 | robl^ | Greenlight: Avaya? or legacy Nortel re-branded as Avaya? |
| 17:14.45 | *** join/#asterisk aruntomar (~Thunderbi@49.248.153.138) |
| 17:16.43 | zamba | [TK]D-Fender: that's actual output, apart from that <local ip>-part |
| 17:17.41 | [TK]D-Fender | zamba: looks like a parse error of some sort. What ver? |
| 17:22.16 | zamba | [TK]D-Fender: 1.8.11.1-1digium1~lucid |
| 17:22.44 | [TK]D-Fender | I'd post up debug of that in a bug-report... |
| 17:31.48 | *** join/#asterisk threesome (~threesome@81.183.251.4) |
| 17:32.43 | Qwell | ~upgrade asterisk |
| 17:32.44 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 17:36.54 | [TK]D-Fender | 1.8.22.0 <- |
| 17:37.04 | [TK]D-Fender | only half a branch old! |
| 17:37.45 | *** join/#asterisk g_r_eek (~g_r_eek@78-89-166.adsl.cyta.gr) |
| 17:47.00 | *** join/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
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| 17:59.33 | jmetro | apparently my queue fix from last night is broken.. again |
| 17:59.44 | jmetro | didnt even change anything but its announcing at the start once more |
| 18:04.45 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 18:09.45 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 18:11.23 | *** join/#asterisk imox (~imox@24.134.18.71) |
| 18:12.58 | rrittgarn | Anyone ever seen a single endpoint (aastra phone in this case) randomly drop off of sip peers completely? As in, Sip show peers -> Yields proper result, a few seconds later, the phone is completely missing from the peer list. I have qualify enabled and all the other aastra phones are staying registered without an issue. nat = force_rport,comedia. I've tried upping registration time, as well as turning up qualifyfreq f |
| 18:13.33 | igcewieling | rrittgarn: not unless you are using Realtime or something |
| 18:13.47 | rrittgarn | which I am |
| 18:14.12 | igcewieling | rrittgarn: any time you do a reload or sip reload then you have to wait for the phone to re-register before it will show up in sip show peers |
| 18:14.28 | igcewieling | (with realtime) |
| 18:14.31 | rrittgarn | I'm aware of that. But I'm not reloading as this endpoint is dropping out of sip peers |
| 18:14.47 | igcewieling | rrittgarn: I cann't imagine how that would happen. |
| 18:15.29 | igcewieling | I assume "dropping out" means "no longer shows up in sip show peers" and not "shows unreachable" |
| 18:15.34 | rrittgarn | Nor can I... which is why I was asking here... |
| 18:15.36 | rrittgarn | correct |
| 18:16.13 | *** join/#asterisk g_r_eek (~g_r_eek@78-18-13.adsl.cyta.gr) |
| 18:16.20 | igcewieling | are you SURE you don't have some script issuing an "asterisk -rx" somewhere? |
| 18:16.25 | igcewieling | like FreePBX or somethign like that |
| 18:16.59 | rrittgarn | yea this is a straight asterisk 11.4.0rc1 built by me. No scripts running. Also, its only this one phone. |
| 18:17.22 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 18:17.23 | *** mode/#asterisk [+o angler] by ChanServ |
| 18:18.20 | rrittgarn | @igcewieling: chan_sip even does the NOTICE: peer is now reachable stuff every time, but I never seen unreachables. It just no longer shows up in sip show peers. |
| 18:19.19 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-swgfnfmwxkwfzffn) |
| 18:25.23 | Katty | raden: ohai <3 |
| 18:29.49 | *** join/#asterisk geeksteve (~geeksteve@195.110.168.123) |
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| 18:53.58 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 19:05.46 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 19:11.56 | robert_ | hey igcewieling |
| 19:12.21 | robert_ | so I don't know how to get Ast to stop using NAT/LAN IP's |
| 19:13.05 | [TK]D-Fender | Do your NAT settings properly |
| 19:13.18 | [TK]D-Fender | And you never showed us your peers & calls as requested |
| 19:18.21 | robert_ | I gave you a complete dump of the console log. I'll get you a list of peers in a moment. |
| 19:25.59 | navaismo | why the billsec passed from dialplan are 1second below the billsec stored in the CDR and provider? i.e: Billsec in dialplan report 19 but cdr & provider report 20 |
| 19:27.54 | *** join/#asterisk Free99 (~Free99@catatc.engr.ccny.cuny.edu) |
| 19:29.06 | Free99 | hey everyone. Having an issue with latency where my peers keep just missing the "qualify" threshold. I know you can set qualify per area in sip.conf, but is there a way to set it globally? I'm using asterisk realtime btw |
| 19:30.56 | [TK]D-Fender | navaismo: Show us your dialplan |
| 19:33.24 | navaismo | LOL ----> https://wiki.asterisk.org/wiki/display/AST/Generating+Billing+Information+from+CEL |
| 19:34.50 | navaismo | [TK]D-Fender, well im trying in the hangup exten --->exten =>h,1,NoOp(The billsec are ${CDR(billsec)}) |
| 19:35.51 | navaismo | Output --> Executing [h@testcontext:1] NoOp("SIP/5005-0000002f", "The billsec are 3") in new stack |
| 19:36.00 | igcewieling | navaismo: [root@daffy-01 ~]# grep endb /etc/asterisk/cdr.conf |
| 19:36.00 | igcewieling | endbeforehexten=yes |
| 19:36.30 | [TK]D-Fender | That's what I was figuring.... |
| 19:36.34 | [TK]D-Fender | "h" delay |
| 19:37.22 | navaismo | Awesome... thanks both of you |
| 19:37.36 | navaismo | now testing |
| 19:39.16 | navaismo | nice now cdr & billsec are the same, provider has 1 second more argh |
| 19:40.31 | Free99 | it was qualifyfreq, if anyone was curious :) |
| 19:47.37 | robl^ | navaismo: rounding differences? X seconds + a fraction of another second. |
| 19:48.14 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 19:48.31 | igcewieling | navaismo: you need to spend more time reading the .sample config files. |
| 19:49.21 | navaismo | just playing with seconds, and trying to confirm that cdr + providers never get the same |
| 19:49.44 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.179) |
| 19:50.01 | navaismo | not like a critical app but thanks all of you |
| 19:50.08 | navaismo | going to eat |
| 19:50.34 | robl^ | some providers always round up. some round up in 8 second increments. so a 5 second call might be billed at 8 seconds |
| 19:53.00 | jmetro | what a gip |
| 19:53.35 | leifmadsen | back in the day it used to be round up to the minute |
| 19:53.41 | leifmadsen | 5 second call == 1 minute |
| 19:56.00 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.179) |
| 19:56.29 | raden | Katty, how are u my dear ? |
| 19:58.21 | robert_ | [TK]D-Fender: is it acceptable if I pastebin the table schema for sip_users, minus any actually identifying information (names replaced with e.g., "SIP user 1", "SIP user 2", etc., callback numbers replaced with e.g., "8130000001") and passwords (replaced with e.g., "secret") along with the INSERTs used to populate the table? |
| 19:59.33 | [TK]D-Fender | mask only passwords |
| 20:08.08 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:28.17 | *** part/#asterisk jkister (~chatzilla@67.200.119.94) |
| 20:31.06 | [TK]D-Fender | And .. another wasted hour |
| 20:31.09 | [TK]D-Fender | time's up... |
| 20:31.12 | [TK]D-Fender | heads home |
| 20:43.59 | *** join/#asterisk blee (~blee@50-89-200-235.res.bhn.net) |
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| 20:47.07 | *** join/#asterisk plUmbro (~gesnaud@cdf35-1-78-230-252-134.fbx.proxad.net) |
| 20:47.14 | plUmbro | Hello there |
| 20:50.30 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:10.32 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 21:17.02 | *** join/#asterisk bpriddy (~bpriddy@ipv4.host.stabbyspazzout.net) |
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| 21:21.17 | rrittgarn | there any way to forcibly hang up a channel that doesn't want to listen to the 'channel request hangup' ? |
| 21:28.31 | newtonr | rrittgarn: "hangup request <channel name>" .. may do the same thing. I'm not sure |
| 21:29.34 | newtonr | yeah thats just an alias |
| 21:30.00 | rrittgarn | doesn't work from cli with either of those. Also tried from an AMI connection to no avail |
| 21:30.44 | newtonr | rrittgarn: pastebin the "core show channel <channel name>" |
| 21:31.40 | [TK]D-Fender | rrittgarn: Use an AMI redirect to toss them off a cliff |
| 21:33.48 | *** join/#asterisk MatBoy (~yamakasi@62.58.32.94) |
| 21:34.35 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 21:35.20 | rrittgarn | apparently if i use the AMI 5 times and then wait, it finally tears down the channel after many autodestruct warnings |
| 21:35.42 | *** part/#asterisk geeksteve (~geeksteve@195.110.168.123) |
| 21:35.51 | rrittgarn | i'll grab full channel info if i get it again. I think a co-worker put a 'g' in a dial somewhere and doesn't hang it up |
| 21:37.27 | *** join/#asterisk janmate (~janmate@chello089173160127.chello.sk) |
| 21:37.33 | *** part/#asterisk janmate (~janmate@chello089173160127.chello.sk) |
| 21:38.41 | *** join/#asterisk monsterco (~monsterco@bas6-toronto47-1279309295.dsl.bell.ca) |
| 21:39.54 | monsterco | hi everyone; can anyone using Aastra phones confirm this: When a 2nd call comes in on an Aastra phone set while there is another call going on, none of the buttons like Transfer, Conference, or DTMF work until 2nd call is picked up or hanged up? |
| 21:42.19 | rrittgarn | @monsterco there's a setting to have incoming calls steal focus or not |
| 21:43.17 | rrittgarn | @monsterco: Log into the phone's web UI -> Preferences -> Switch UI Focus To Ringing Line - Uncheck the box |
| 21:43.56 | monsterco | rrittgarn - thanks a lot man - let me try that now |
| 21:44.04 | monsterco | thought this won't get fixed at all |
| 21:45.20 | rrittgarn | I actually am an Aastra Reseller, and have an open ticket on some other firmware issues some of my customers have found... |
| 21:46.07 | monsterco | rrittgarn - reseller too but through distributor and never bothered to do tickets - do you encounter the web server issue from time to time? |
| 21:46.26 | monsterco | phone Web GUI not being responsive and then opening fine with HttpS |
| 21:47.13 | rrittgarn | <PROTECTED> |
| 21:49.47 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-jqufhjycpzacowos) |
| 21:50.43 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 21:50.46 | monsterco | rrittgarn - that has been going on for ever - they always deny it's happening but it's happening with everyone in the field |
| 21:51.05 | monsterco | probably the web server base code is somehow bad or hardware issue but its really annoying |
| 21:51.11 | monsterco | the focus worked - thanks a lot |
| 21:51.32 | monsterco | it's funny how there are so many things these phones can do that is not so obvious |
| 21:52.38 | rrittgarn | yeah... i've gotten into a lot of intricate setups and these phones just seem to handle it all without an issue. not to mention the XML Stuff... we do custom apps for our clients on these phones and they work really well too |
| 21:53.03 | monsterco | what do the apps do mostly? |
| 21:54.46 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 21:55.24 | rrittgarn | the most commonly used is the display users on park by caller ID and who they are parked by. Have one that a client wants a panic button that dials 911, locks the building down, etc... that one should be fun... then other basics too like multiple blfs based on non-phone stuff and launching CRM specific things or picking which channels to listen to, etc.... pretty flexible as far as what you can make them do |
| 21:56.51 | monsterco | "locks channels to listen to" ? |
| 21:57.51 | rrittgarn | hmm? I had 'Locks down the building" as well as visually pick when channels to listen to (via chanspy) |
| 21:57.55 | monsterco | I always fantasize about a radar like dashboard with multiple screens that would show me status of all calls so NOC can barge in on a call and see quality of the bandwidth/call and other info related to all customers and stations |
| 21:58.28 | rrittgarn | wouldn't be terribly hard |
| 21:58.47 | monsterco | So, the screen shows all channels that are live and then you use 57i screen to list those channels and to listen to them? |
| 21:58.55 | rrittgarn | yeah |
| 21:59.03 | monsterco | that's neat |
| 21:59.43 | monsterco | yeah, it won't be hard but just the time to do it btw all the other 1000 priority tasks :) |
| 21:59.59 | monsterco | do you use redirector from Aaastra? |
| 22:02.29 | rrittgarn | Not sure. I've used a few tools from them, that one doesn't sound familiar though... |
| 22:03.40 | rrittgarn | ah looked it up. No, we wrote our own provisioning server that handles Aastra, Cisco, and Polycom |
| 22:04.51 | monsterco | Well, redirector allows you to ship phone direct from distributor and redirector points to your IP based on the MAC address that was sold |
| 22:04.59 | monsterco | and then you can push your own provisioning |
| 22:05.17 | monsterco | more suitable for a non-vpn environment I guess |
| 22:05.29 | monsterco | but I think it's $2000 per year so - def not worth it |
| 22:13.35 | jmetro | depends on your volume and worth of convenience |
| 22:14.11 | jmetro | pay for 100 phones to your office, 100x the cost of provisioning, and 100x travel expenses to clients could definitely be worth 2000$ |
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| 22:48.01 | picard276 | hey guys i was wondering as t oo |
| 22:48.10 | picard276 | how to do a sip debug but only show the logs from a certain IP |
| 22:48.17 | picard276 | when i do sip set debug on |
| 22:48.21 | picard276 | there are so many messages it really counter productive |
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| 22:49.08 | picard276 | ? any ideas |
| 22:49.20 | WIMPy | Use your tab key |
| 22:49.39 | WIMPy | Then yu wil find that you can use 'sip set debug ip ...' |
| 22:51.13 | picard276 | thanks |
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| 22:57.13 | DBordello | Yay! I have my first PBX :) |
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| 23:37.34 | DBordello | How do I list what modules are installed and loaded? |
| 23:40.23 | robl^ | DBordello: try: "module show" |
| 23:41.32 | DBordello | great, thanks. |
| 23:41.55 | DBordello | I searched for a good while for that. The documentation for Asterisk seems pretty weak. What would be the best way for me to figure that out myself? |
| 23:42.22 | robl^ | try "help" in the asterisk console ;-) |
| 23:42.58 | WIMPy | ~book |
| 23:42.59 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 23:43.58 | DBordello | robl^, that is a good resource ;) |
| 23:44.57 | robl^ | when do we get a new edition of the book for Asterisk 11 -- my first 3 editions are starting to want another to sit beside them on the shelf. |
| 23:45.52 | robl^ | ohh. there is a 4th edition in review. I missed that. |
| 23:52.53 | DBordello | I might check out the 4th edition then |