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| 00:04.26 | navaismo | as you can see im lost too |
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| 00:12.01 | lwizardl | whats the best way to set it so that each room is unique. basically a press 4 for Joe, 5 for Jane, etc |
| 00:12.54 | WIMPy | We don't configure roomes, except for conference rooms. But you can configure phones. |
| 00:13.04 | WIMPy | I think you should start with the |
| 00:13.07 | WIMPy | ~book |
| 00:13.08 | 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 |
| 00:13.31 | lwizardl | k thanks |
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| 01:00.09 | volga629 | Hello Everyone, is syntax for exten => s,n,Set(FDST=${STRREPLACE(${PDST},find,replace,1)}) correct ? |
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| 04:36.41 | *** join/#asterisk Aivaras (~Aivaras@295864.s.dedikuoti.lt) |
| 04:37.23 | Aivaras | Hey, I just seted up Asterisk and two extensions. I can call from 1 to 2, but not from 2 to 1. Where to look for a problem? |
| 04:37.31 | *** join/#asterisk AliRezaTaleghani (~Thunderbi@unaffiliated/AliRezaTaleghani) |
| 04:40.24 | Penguin | ~pb |
| 04:40.24 | 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. |
| 04:41.03 | Aivaras | Penguin, is it for me? :/ I don't know what to paste |
| 04:41.05 | Penguin | Pastebin your dialplan, at least. |
| 04:41.37 | Aivaras | http://pastebin.com/miWNNi4e |
| 04:42.03 | Aivaras | I can call to 6002 from 007, but I can't call from 6002 |
| 04:42.23 | Penguin | 007 is an extension. You don't call FROM it. |
| 04:43.05 | Penguin | You call to extensions from phones. |
| 04:43.18 | Penguin | Which phone is giving you a problem? |
| 04:43.30 | Aivaras | then I can't call from aivaraspc to aivaras |
| 04:44.34 | Aivaras | I tried ekiga (pc software) and zoiper (android) |
| 04:44.44 | Penguin | Pastebin the sip configs for both phones. |
| 04:45.35 | Penguin | If nothing obvious appears there, we'll go to a sip debug. |
| 04:46.24 | Aivaras | I can't export settings from android, but it's just username, password and ip of server. |
| 04:46.46 | Penguin | I'm talking about the sip config from sip.conf. |
| 04:47.11 | Penguin | Each phone should have one entry there. |
| 04:47.48 | Aivaras | http://pastebin.com/Pi6e5C0X |
| 04:47.52 | Aivaras | everything else is default |
| 04:48.52 | Penguin | I suppose we should look at the sip debug now. sip set debug on |
| 04:49.11 | Penguin | Make the call that fails. sip set debug off |
| 04:49.20 | Penguin | Pastebin everything from start to finish. |
| 04:50.04 | Aivaras | it works now :/ |
| 04:50.25 | Penguin | That was an easy fix. |
| 04:50.33 | Aivaras | but I did nothing. |
| 04:50.53 | Penguin | It is possible that aivaras was not registered at the time you tried to call it. |
| 04:51.01 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 04:51.11 | Aivaras | I did that reload thing |
| 04:51.16 | Penguin | sip reload? |
| 04:51.20 | Aivaras | yeah |
| 04:51.25 | Aivaras | end then extensions reload |
| 04:51.26 | Aivaras | oh |
| 04:51.33 | Aivaras | dailplan reload or something |
| 04:51.45 | Penguin | Yeah, dialplan reload. |
| 04:52.04 | Aivaras | yeah it was that. So all I did was debug on and it started to work. |
| 04:52.09 | Penguin | It's possible that your extensions had not been reloaded after you made a change. |
| 04:52.31 | Penguin | You have to reload it any time you edit it. |
| 04:52.37 | Aivaras | btw, what codec I should use? |
| 04:52.45 | Aivaras | I understand that. |
| 04:52.46 | Penguin | That depends. |
| 04:53.08 | Aivaras | I have 40 mbps server with 70ms latency. |
| 04:53.11 | Penguin | The typical answer could be ulaw or alaw... |
| 04:53.23 | Aivaras | because it's 2k km away from location I am using it. |
| 04:53.37 | Penguin | But that might not be the best codec for your particular application. |
| 04:54.21 | Penguin | If your calls are going to/from the PSTN, go with alaw. |
| 04:55.18 | Aivaras | pstn - it's something with real phone networks? |
| 04:55.39 | Penguin | You can even use alaw between the PSTN and your box, and use something with better quality on the phones themselves for when you talk between the phones (if the phones support HD audio). |
| 04:55.49 | Penguin | ~pstn |
| 04:55.49 | infobot | extra, extra, read all about it, pstn is Public Switched Telephone Network, or "please stop the nonsense" |
| 04:56.06 | Penguin | PSTN, or real phones. |
| 04:57.09 | Penguin | A regular old phone doesn't support high quality audio. |
| 04:57.27 | Aivaras | yeah. I think I will be using just android phones over wifi |
| 04:57.45 | Penguin | I really like CSipSimple on Android. |
| 04:58.02 | Penguin | It seems better than the others I have tried... and I tried as many as I could find. |
| 04:58.24 | Penguin | Zoiper is great on a Windows PC, but not as good on Android. |
| 04:58.52 | Aivaras | it looks neat. |
| 04:59.05 | Penguin | CSipSimple has decent integration with the native phone, too. |
| 04:59.20 | Aivaras | I will try it :) |
| 04:59.21 | Penguin | Good dialer integration, at least. |
| 04:59.55 | Penguin | I have mine set up to ask me which phone to complete the call with when I try to make a call. |
| 05:00.17 | Penguin | If the SIP phone is registered, it prompts me; if it is not registered, it just uses the wireless phone. |
| 05:00.56 | Aivaras | in my case it's more "try new things" then I really need it :D |
| 05:01.13 | Penguin | Experimenting is most of the fun. |
| 05:01.24 | Penguin | It's boring just using the phone to talk on. |
| 05:02.05 | ChannelZ | That's why there's phone sex. |
| 05:02.17 | Penguin | Phone sex, you say? |
| 05:02.57 | Penguin | Now I'm curious if they make phone-shaped condoms. |
| 05:03.12 | ChannelZ | Saran wrap |
| 05:03.22 | Aivaras | I think you can use regular ones. :D |
| 05:03.48 | Penguin | That would be easier than asking Motorola to make a phone shaped like, well, you know. |
| 05:04.06 | ChannelZ | say it! |
| 05:04.28 | Penguin | A corndog! |
| 05:04.35 | Aivaras | what is best sip app for linux? |
| 05:04.50 | Aivaras | is ekiga usable? |
| 05:05.16 | Penguin | I used to think it was twinkle, but lots of people like a couple others that I can't think of the names of. |
| 05:05.30 | Penguin | Blink, jitsi? Something like that. |
| 05:06.13 | Penguin | For Windows, I always went to Zoiper. For my desktop, I always used twinkle. |
| 05:06.45 | Penguin | Ekiga was usable, but not desirable. |
| 05:06.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.204) |
| 05:07.26 | Aivaras | hm.. now I can't call from aivaras to aivaraspc :D |
| 05:07.31 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-sfnllezhtugwsgeb) |
| 05:07.37 | Penguin | sip show peers |
| 05:07.46 | Penguin | Is aivaraspc registered? |
| 05:08.00 | Aivaras | yeah |
| 05:08.07 | Aivaras | unmonitored, online |
| 05:08.24 | Penguin | sip debug, I suppose. |
| 05:08.41 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 05:10.07 | Penguin | So, I have to share something I heard earlier. It was amusing to me... |
| 05:11.32 | Aivaras | cat I save debug info as log? It sucs to get it realtime |
| 05:11.38 | Penguin | There were some guys talking on the ham radio (75 meters), one said something about making a note to himself about a corndog in the microwave and something about ranch dressing and mustard. (I'm sure the details of his statement weren't too important.) |
| 05:12.22 | Penguin | Another guy says, "You've got a soldering iron... why are you using a microwave to cook a hotdog?" |
| 05:13.45 | Penguin | By what method are you connected to the asterisk console? |
| 05:13.51 | Aivaras | terminal |
| 05:13.56 | Aivaras | I have log now |
| 05:13.57 | *** join/#asterisk aruntomar (~Thunderbi@49.248.153.148) |
| 05:14.14 | Aivaras | http://pastebin.com/ysDnr78g |
| 05:14.26 | Aivaras | just made asterisk -r > log |
| 05:16.16 | Penguin | I see a 404 not found in there. |
| 05:16.29 | Aivaras | it says 82.30.211.63:5060 - do I need to fwd that port to my device? :/ |
| 05:16.33 | Penguin | Are your two phones both on the same LAN as your Asterisk box? |
| 05:16.39 | Aivaras | no |
| 05:16.49 | Aivaras | phones on lan, server away |
| 05:17.11 | Penguin | You will have to forward 5060 inbound to your asterisk IP address if it is behind a NAT. |
| 05:17.41 | Aivaras | to server? |
| 05:17.48 | Penguin | Be sure to tell asterisk your phones are behind NAT if they are. |
| 05:17.48 | Aivaras | all my server ports are open |
| 05:18.57 | Penguin | I didn't see any private addresses in the debug, so that could be a problem if your phones are behind a NAT. |
| 05:19.44 | Aivaras | if I settup a proxy on server and usi on phones? |
| 05:20.07 | Penguin | I don't know why you would want to do that. |
| 05:20.21 | Aivaras | That would solve connection problems? |
| 05:20.32 | Penguin | Sounds like it would complicate things. |
| 05:20.56 | Aivaras | so what should I do? |
| 05:21.27 | Penguin | A typical configuration would be to tell asterisk that the phones are behind NAT and make sure the phones have any NAT fix disabled. |
| 05:21.44 | Aivaras | hm |
| 05:22.03 | Aivaras | I can call to exetensions, but I can't call to "aivaras" |
| 05:22.31 | Aivaras | I mean, I can't do "call back" |
| 05:22.46 | Penguin | If asterisk doesn't know how to get a call to it, that makes sense. |
| 05:23.50 | Aivaras | Ok, everything seams to be working :) |
| 05:23.52 | Aivaras | Thanks |
| 05:24.09 | Penguin | I would add nat=yes to both of the phones' sip entries. |
| 05:24.16 | Penguin | Save, then sip reload. |
| 05:25.06 | Aivaras | done |
| 05:25.08 | Penguin | That way asterisk can be aware of private addresses within packets to/from those devices. |
| 05:25.14 | Aivaras | now call back should work? |
| 05:25.30 | Penguin | Try it. |
| 05:25.46 | Aivaras | no |
| 05:26.17 | Penguin | I'm not really sure what "call back" is. It sounds like you are describing a button on the phone. |
| 05:26.46 | Aivaras | well.. it is :D |
| 05:27.02 | Penguin | That might be a feature that isn't compatible with asterisk. Or it might be something that you would need to take extra steps in dialplan for it to work. |
| 05:27.13 | Aivaras | I see that "aivaras" called me on "aivaraspc", but if I press call back, it says 404 |
| 05:27.19 | Aivaras | oh |
| 05:27.21 | Aivaras | ok then |
| 05:27.53 | Penguin | It might be trying to call the other phone directly, or it might just be telling asterisk the wrong thing. |
| 05:28.27 | Penguin | SIP headers will often have goofy stuff in there that you need to read carefully to debug something like that. |
| 05:29.11 | Penguin | Actually... the call probably isn't going to the phone's associated extension. |
| 05:29.38 | Penguin | So when the one phone tries to call a phone by name instead of extension number, that fails. |
| 05:30.02 | Penguin | The SIP debug would probably reveal that if you look in the INVITEs. |
| 05:30.24 | Aivaras | yea, but since I can call exenssions - it's ok for me :D |
| 05:30.46 | Penguin | There would be a couple ways you could fix that. |
| 05:32.04 | Penguin | One is create a duplicate extension for the phone, but switch the number with the name. So then you would have extension 007 as well as extension aivaras, both would dial SIP/aivaras |
| 05:32.24 | Aivaras | I can try that |
| 05:32.33 | Penguin | Another is to not press the call back button. :) |
| 05:32.38 | Aivaras | lol |
| 05:33.41 | Penguin | I'm sure I have worked through this before, but I just don't recall what ways I would have done it. |
| 05:34.02 | Penguin | Sometimes trying to be creative just causes trouble. |
| 05:34.41 | Aivaras | it gives 503 then hitting redail |
| 05:35.40 | Penguin | You can spend a long time making everything work. |
| 05:36.32 | Penguin | Debug each problem, figure out how to overcome it. |
| 05:36.32 | Aivaras | now I would like to make web interface for adding new users :) |
| 05:36.57 | Penguin | There are interfaces for that already, but I can't recommend you use them. |
| 05:37.09 | Aivaras | I want make one myself. |
| 05:37.18 | Penguin | vim is pretty easy to use, so I wouldn't bother. |
| 05:38.05 | Aivaras | yeah, but I can't give you my server root password, but I could let you register on website :) |
| 05:38.13 | [TK]D-Fender | Looking for aivaraspc in users (domain 256.lt) SIP/2.0 404 Not Found |
| 05:38.22 | [TK]D-Fender | Error looks mre than clear... |
| 05:38.29 | [TK]D-Fender | No dialplan match just like it says |
| 05:38.57 | Penguin | If you know python, you could write something to edit your sip.conf or an included file and run sip reload when it is done. |
| 05:39.27 | Aivaras | yea. |
| 05:39.54 | Aivaras | [TK]D-Fender, http://pastebin.com/yMRCbw4g |
| 05:40.05 | Penguin | There's also the option of using real time configs and editing that with some other homemade app. |
| 05:40.26 | [TK]D-Fender | Aivaras: Which confirms it |
| 05:41.05 | Penguin | SIP/007 and SIP/6002 don't exist, so be careful with that. |
| 05:41.45 | Aivaras | so: exten=>aivaras,1,Dial(SIP/aivaras,20) ? |
| 05:42.22 | Aivaras | it works!!! |
| 05:42.35 | Aivaras | [TK]D-Fender, thanks! |
| 05:42.50 | Aivaras | and many thanks to Penguin :D |
| 05:43.04 | Penguin | I was thinking something like this: http://pastebin.com/kR8kksyp |
| 05:43.06 | [TK]D-Fender | Aivaras: I advise againstr trying to be "clever" and using "names" for your dilaplan extensions. |
| 05:43.47 | Penguin | It was my first thought for making the silly "call back" button work. |
| 05:43.48 | Aivaras | [TK]D-Fender, I was just following wiki, there was 6001 and 6002 |
| 05:43.50 | [TK]D-Fender | Aivaras: Because it seems nifty at first and then when youa re using some crappy softphone on a PC and you're stuck with a numeric keypad.... and can't type it in easy that life will suck. |
| 05:44.08 | [TK]D-Fender | Aivaras: Same for IVR's you'll set up that the outside world may reach. |
| 05:44.32 | [TK]D-Fender | Aivaras: Forget "wiki". Think "this what what I want to DIAL, and this is what I want it to DO". |
| 05:44.59 | Aivaras | I wanted to see it working. Tweeking is later :D |
| 05:45.01 | [TK]D-Fender | airAnd remember you are dialing an EXTENSION, not a SIP DEVICE when you place a call. |
| 05:45.15 | Penguin | I'm suddenly reminded of how parts of relevant conversation seem to slip past [tk]d-fender so often. |
| 05:45.17 | [TK]D-Fender | Aivaras: Wyat your dialplan does is another matter. |
| 05:45.53 | Penguin | He set up numeric extension to dial named devices. No problem there. |
| 05:46.08 | [TK]D-Fender | [01:41]Aivarasso: exten=>aivaras,1,Dial(SIP/aivaras,20) ? |
| 05:46.09 | [TK]D-Fender | [01:42]Aivarasit works!!! |
| 05:46.11 | [TK]D-Fender | ^^ |
| 05:46.14 | [TK]D-Fender | he just did the named ones |
| 05:46.18 | Aivaras | and setted up "name" extensions just to use call back button |
| 05:46.22 | Penguin | But then his soft phone wasn't too bright and the Call Back button wanted to call the phone back by name instead of number. |
| 05:46.44 | Aivaras | http://pastebin.com/aBjnCu6f |
| 05:46.52 | Penguin | Now that you're caught up, we can move on. |
| 05:47.19 | [TK]D-Fender | Aivaras: Show us that inital call that you say doesn't register properly. |
| 05:47.41 | Aivaras | now everything works. |
| 05:47.47 | Penguin | It was the one where it said looking for aivaras in users. |
| 05:47.58 | [TK]D-Fender | still a twisted way to call back.. something looks "off" |
| 05:48.14 | Aivaras | I think it's because of app. |
| 05:48.19 | Penguin | That was a result of the phone's stupod call back button trying to call the device back by name instead of extension number. |
| 05:48.39 | [TK]D-Fender | I see why |
| 05:48.42 | Penguin | Which soft phone is that? |
| 05:48.56 | [TK]D-Fender | http://pastebin.com/Pi6e5C0X <-- you didn't set the CALLERID so you could have the NAME and NUMBER |
| 05:48.58 | Aivaras | zoiper |
| 05:49.06 | [TK]D-Fender | Aivaras: Fix your peers |
| 05:49.22 | Aivaras | how? :D |
| 05:49.32 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 05:49.41 | [TK]D-Fender | callerid="john doe" <101> |
| 05:49.49 | Penguin | without the quotes, of course. |
| 05:50.04 | [TK]D-Fender | Quotes are perfectly valid there |
| 05:50.10 | [TK]D-Fender | Always have been |
| 05:50.22 | [TK]D-Fender | And are not interpreted literally into the name |
| 05:50.24 | Penguin | Maybe if you like double-double quotes. |
| 05:50.30 | Penguin | They used to be literal. |
| 05:50.38 | Penguin | I doubt it changed in the past year. |
| 05:50.39 | [TK]D-Fender | they are NOT brought into the string on this |
| 05:50.42 | Aivaras | callerid="Aivaras Kivilius" <007> ? |
| 05:50.43 | [TK]D-Fender | never for callerid |
| 05:50.52 | [TK]D-Fender | Aivaras: Yes |
| 05:51.20 | Penguin | I would encourage you to read back over years of talking about the callerid value. |
| 05:51.23 | [TK]D-Fender | it's been this way since before Asterisk broke it's first whole number version... |
| 05:51.41 | [TK]D-Fender | I've been doing this for just under a decade.... |
| 05:51.49 | [TK]D-Fender | I don't need to look back. I was there |
| 05:51.57 | Penguin | What a coincidence. I was, too. |
| 05:52.12 | [TK]D-Fender | Well I also happen to be right on this :) |
| 05:52.22 | Penguin | We've talked about it many times. |
| 05:52.47 | [TK]D-Fender | Quote on Callerid? nothing noteworthy... |
| 05:52.47 | Penguin | You end up with two sets of quotations if you quote the name in the callerid value. |
| 05:53.39 | Penguin | It should be callerid=Aivaras Kivilius <007> |
| 05:54.46 | [TK]D-Fender | Should work as well |
| 05:55.15 | Aivaras | And I can remove those double extension lines? |
| 05:55.20 | [TK]D-Fender | yes |
| 05:55.43 | Aivaras | 404 |
| 05:55.48 | Penguin | I was going to lay all the blame on the soft phone on that one. |
| 05:55.54 | Penguin | Remember to dialplan reload? |
| 05:56.01 | Aivaras | yes |
| 05:56.06 | [TK]D-Fender | and "sip reload"? |
| 05:56.16 | Aivaras | yes |
| 05:56.27 | [TK]D-Fender | well then you should see the number on your new call out |
| 05:56.29 | Aivaras | oh |
| 05:56.33 | Penguin | Of course you did dialplan reload... that was how you got back to having a 404. I'm tired. |
| 05:56.33 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 05:56.34 | Aivaras | yes |
| 05:56.36 | Aivaras | I do |
| 05:56.44 | [TK]D-Fender | I hope you didn't just try to dial that NAME back from the same entry as before... |
| 05:56.52 | Aivaras | I did. :D |
| 05:56.53 | Aivaras | my bad. |
| 05:57.02 | [TK]D-Fender | NEW call :p |
| 05:57.08 | Aivaras | Now I see ID and I can call it back |
| 05:57.22 | Aivaras | Do I need extension line at all? |
| 05:57.45 | [TK]D-Fender | of course... |
| 05:57.52 | [TK]D-Fender | that number has to do something, doesn't it? |
| 05:57.52 | Aivaras | or sip just changes id (how it looks) and has nothing to do with extension? |
| 05:57.55 | Penguin | I was hoping they didn't break zoiper. I used to really like it. |
| 05:58.12 | [TK]D-Fender | They didn't break Zoiper as far as we've seen. |
| 05:58.26 | Penguin | Yeah. I just thought they did. |
| 05:58.30 | [TK]D-Fender | Improper callback info was provided due to lack of configuration |
| 05:58.35 | Penguin | I agree. |
| 05:59.44 | Aivaras | what "Aivaras Kivilius" does in line callerid=Aivaras Kivilius <007> ? |
| 05:59.57 | Penguin | I never even considered the callerid on that call back button. I had no idea what it was atually doing other than trying to call back the phone that called it. |
| 06:00.06 | Penguin | That is the callerid name. |
| 06:00.22 | Aivaras | but I can't see it on app. |
| 06:00.27 | Aivaras | I just see 007 |
| 06:00.28 | Penguin | callerid=Your Name <your phone number> |
| 06:00.49 | Penguin | Maybe the app doesn't show the callerid name? |
| 06:00.50 | [TK]D-Fender | You should be able to see name and number |
| 06:00.52 | Penguin | I thought it did. |
| 06:00.56 | [TK]D-Fender | possible |
| 06:01.13 | Penguin | It has been a while since I used it, but I thought it showed the name and number. |
| 06:02.06 | Aivaras | http://i.imgur.com/JfWLHgx.png |
| 06:02.13 | Penguin | I also haven't used a version of it made in the last two years. |
| 06:02.55 | Penguin | Oh. That doesn't surprise me. |
| 06:03.11 | Penguin | On the incoming call, you should see name and num. |
| 06:03.54 | Penguin | On the return call button, they probably just didn't bother putting caller name on it. |
| 06:04.13 | Aivaras | no name on incoming call to |
| 06:04.19 | Penguin | :/ |
| 06:04.21 | Penguin | pewp |
| 06:04.32 | Penguin | Try CSipSimple. |
| 06:04.35 | Aivaras | ok |
| 06:05.28 | Penguin | Even though I use gingerbread, the soft phone should still work about the same way for you. |
| 06:06.14 | Penguin | Are you using Fro Yo? |
| 06:06.34 | Aivaras | ? |
| 06:06.48 | Penguin | What is your android version? |
| 06:06.53 | Aivaras | 4.3 |
| 06:07.10 | Penguin | Jelly Bean |
| 06:07.14 | Aivaras | yeah |
| 06:08.03 | Penguin | I don't know what I was thinking. |
| 06:08.11 | Aivaras | http://i.imgur.com/acdDKA7.png |
| 06:08.59 | Penguin | Got a name on that one. |
| 06:09.30 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 06:10.15 | Penguin | I use my native dialer and enabled dialer integration in csipsimple. |
| 06:10.22 | Penguin | It's acceptable. |
| 06:10.38 | Aivaras | Yeah, I seen this option. Will try later. :) |
| 06:11.09 | Penguin | There are so many sip phones, you can spend all night trying them all. |
| 06:11.28 | [TK]D-Fender | And on that note... it's bed time... |
| 06:11.32 | [TK]D-Fender | back later |
| 06:11.36 | Penguin | past bed time! |
| 06:11.39 | Aivaras | it's 7:11 there. Night was spent. :D |
| 06:11.46 | Aivaras | yeah. :D |
| 06:11.52 | Penguin | 0111 here. |
| 06:11.59 | Penguin | I should have left over 1 hour ago. |
| 06:12.24 | Penguin | I usually don't even come on here much anymore because I spend too long here. |
| 06:12.35 | Penguin | HOURS, not minutes. |
| 06:13.20 | Penguin | I guess that means I'll drop out, too. |
| 06:13.27 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 06:13.49 | Aivaras | wait.. it's -6 from GMT. |
| 06:13.49 | Penguin | Have fun experimenting with soft phones. |
| 06:13.53 | Aivaras | Are you in USA? |
| 06:13.57 | Penguin | Yes. |
| 06:15.55 | Penguin | I don't think I have any Lithuania radio contacts. You don't happen to have an amateur radio, do you? |
| 06:16.45 | Aivaras | I don't and I am in UK. My server is in Lithuania. :D |
| 06:16.50 | Penguin | oh |
| 06:17.01 | Aivaras | And I am lithuanian. But now in UK. |
| 06:17.12 | Penguin | Ah well. I'll try again next time. :) |
| 06:17.33 | Aivaras | what frequencies that radios are working in? |
| 06:19.36 | Penguin | 1.8, 3.5, 5.3, 7, 10.1, 14, 18, 21, 24, 28, 50 MHz |
| 06:20.11 | Penguin | 160, 80, 60, 40, 30, 20, 17, 15, 12, 10, and 6 meter bands |
| 06:20.38 | Aivaras | lol. I prefer 2.4/5GHz waves :D |
| 06:20.54 | Penguin | That's the radio I would have used to talk to .lt if you were there with a radio. |
| 06:21.20 | Penguin | I also have other bands/freqs on other radios. |
| 06:21.54 | Penguin | 2.4/5 GHz doesn't travel around the globe very well. |
| 06:21.55 | Aivaras | I have that cheap walkie-talkie - Binatone Terrain 750 |
| 06:22.17 | Penguin | Those freqs kind of go in a straight line. |
| 06:22.38 | Aivaras | we used them in skiing trip. |
| 06:22.43 | *** join/#asterisk Freeaqingme (~Freeaqing@2a03:f80:ed15:37:235:56:92:7100) |
| 06:22.49 | Penguin | No reflection back to Earth from the ionosphere. |
| 06:23.14 | Penguin | Maybe bounce it off of the moon or some other satellite. |
| 06:23.37 | Aivaras | I think they don't travel that much at all. |
| 06:23.57 | Penguin | With enough power, maybe? |
| 06:24.17 | Aivaras | well that would be nice word-wide wifi jammer |
| 06:25.03 | Penguin | The wifi freqs overlap with US amateur radio freqs, so I can run 1500W on channels 8-11. |
| 06:25.26 | Aivaras | :O |
| 06:25.57 | Aivaras | 1.5kW is a lot of power |
| 06:26.28 | Aivaras | let me check, but I think my wifi is like 500mW or so :D |
| 06:27.30 | Aivaras | 500mW on low power setting and 1W on max :D |
| 06:27.49 | Penguin | No, I told you wrong. Channels 1-8 are in US amateur band. |
| 06:28.49 | Penguin | I can run 1500W on channels 1-8. |
| 06:28.53 | Aivaras | I am on 3 (2.422 GHz) |
| 06:29.25 | Penguin | That band is 2390-2450 MHz. |
| 06:29.42 | Aivaras | in UK some 4G networks are in TV band. |
| 06:30.29 | Penguin | The other band is 5650-5925 MHz. |
| 06:30.39 | Aivaras | 800Mhz |
| 06:31.08 | Aivaras | https://at800.tv/guide/4g-filters-buying-guide/ so we got some sort of filters |
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| 06:32.06 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 06:33.13 | Penguin | 5GHz wifi channels 132-165 |
| 06:35.37 | Penguin | I have no idea how well a 1.5 kW wifi signal will travel. |
| 06:36.53 | Penguin | If I had the amplifiers and the radios, I would try to find out. |
| 06:37.10 | Aivaras | You should |
| 06:37.16 | Aivaras | That is really interesting. |
| 06:37.39 | Aivaras | have you heard about google's internet from baloons ? |
| 06:38.02 | Penguin | They call it line of sight, but I have a feeling the actual sight of it might not be imperative with that much power at both ends of the link. |
| 06:38.11 | Aivaras | http://www.google.com/loon/#utm_source=google&utm_medium=cpc&utm_campaign=Global_semBK |
| 06:38.17 | Penguin | I heard about it, but I don't know how true it is. |
| 06:38.51 | Aivaras | Well. It's google. It's true. :D |
| 06:39.22 | Penguin | 802.11b over 200 miles? |
| 06:40.01 | Penguin | Need tall buildings to mount the antenna. |
| 06:41.01 | Penguin | Would the data degrade over long distances as long as the power level could get the signal to the other side? |
| 06:41.36 | Aivaras | They don't use 802.11 |
| 06:41.56 | Aivaras | You will need some sort of box to connvert signal |
| 06:42.07 | Penguin | I'm still talking about the overlap in the 802.11 and amateur bands. |
| 06:42.21 | Aivaras | oh |
| 06:42.35 | Aivaras | yes, it will degrade. |
| 06:42.53 | Penguin | Get a couple WRT54G plastic routers and connect them to some high power amplifiers. |
| 06:43.53 | Penguin | Data transmissions are hell on amplifiers, so we'll need some good cooling. |
| 06:44.32 | Penguin | It would be fun to try, but it's not practical for me to try it. |
| 06:44.56 | Penguin | Too much money involved for equipment. |
| 06:45.03 | Penguin | Plus... |
| 06:45.08 | Penguin | I would rather be sleeping. |
| 06:45.27 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 06:45.32 | Penguin | So I think I will go do that right now. |
| 06:46.12 | Penguin | Il y a plus d'une façon de peler un chat. |
| 06:46.32 | Penguin | Bye! |
| 06:46.38 | Aivaras | Bye and thanks! |
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| 07:52.54 | bulkorok | hi |
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| 09:39.25 | adnc | hello, is it possible to see if the password for a iax2 trunk is given wrong and the peer is not up because of a password mismatch? |
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| 10:09.25 | hanuman | hi |
| 10:10.06 | hanuman | how can i acheive pure secure asterisk |
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| 10:51.55 | eject_ck | Asterisk behind router NAT, I've forwarded 5060, 10000 - 20000 (RTP) |
| 10:52.02 | eject_ck | is it enough ? |
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| 11:05.04 | eject_ck | I can't get it workin g:( |
| 11:05.34 | eject_ck | i've forwarded all udp ports to asterisk server with no luck |
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| 11:43.49 | Itsm | Hi folks, anyone care to help with some asterisk variables issue? |
| 11:45.05 | wdoekes | ~ask |
| 11:45.05 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 11:45.56 | Itsm | lol, alright. |
| 11:45.57 | Itsm | I'm trying to edit the record macro |
| 11:46.05 | Itsm | And it's going well, I've organised it a format that I want it to save as |
| 11:46.14 | Itsm | But.. I cannot find the correct variable, to make it output the extension that answered to an incoming call |
| 11:46.24 | Itsm | http://pastebin.com/Geew6B0C |
| 11:46.36 | Itsm | the goal is, that it will output From-(incoming number) To-(extension that picked up) |
| 11:46.58 | adnc | wdoekes: even then answer is not garantied |
| 11:49.08 | wdoekes | macros? smells old. |
| 11:50.53 | wdoekes | I have no idea where your macro is called from, so I cannot tell what variable would hold the To-info you're looking for |
| 11:51.32 | adnc | is it possible to see if the password for a iax2 trunk is given wrong and the peer is not up because of a password mismatch? |
| 11:52.22 | wdoekes | adnc: it is possible, but the correlation isn't made automatically |
| 11:53.19 | adnc | wdoekes: and how? |
| 11:53.34 | wdoekes | adnc: I suppose the security event log should tell you that a registration attempt failed |
| 11:54.05 | adnc | but registration attemts can also fail because of other reasons (qualify problems, firewall problems) |
| 11:55.19 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 11:55.31 | wdoekes | <PROTECTED> |
| 11:55.32 | wdoekes | <PROTECTED> |
| 11:55.39 | wdoekes | <PROTECTED> |
| 11:55.40 | wdoekes | <PROTECTED> |
| 11:55.46 | wdoekes | <PROTECTED> |
| 11:55.47 | wdoekes | <PROTECTED> |
| 11:55.54 | wdoekes | I don't see the security event log though |
| 11:56.41 | wdoekes | so you're probably stuck parsing the regular notice log |
| 11:57.25 | wdoekes | oh.. you we're talking about outbound registrations |
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| 11:59.56 | adnc | yes, but I was looking for a simple command on the cli |
| 12:00.20 | adnc | I can't see which one it could be by reading asterisk code above |
| 12:04.01 | wdoekes | <PROTECTED> |
| 12:04.01 | wdoekes | <PROTECTED> |
| 12:04.06 | wdoekes | that would probably be the one then |
| 12:04.13 | wdoekes | and no.. no cli command.. log files |
| 12:06.30 | wdoekes | iax2_show_registry() => regstate2str => "Rejected" |
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| 14:31.45 | igcewieling | Our PRI is down, circuit is fine, D-channel won't come up. Tried Asterisk and adtran boxes, no d-channel comes up on either one. so after 3 days of going back and forth with the carrier they finally dispatch. After a while my boss gets a call from the tech. ""can you guys plus the PRI back into your PBX? I'm having a problem with my test set thats keeping the D channel from coming up"" |
| 14:32.29 | tzafrir | http://www.theguardian.com/money/blog/2013/jul/29/courier-scam-lose-money-bank-cards - does this make sense (the bit about call not disconnected)? |
| 14:35.17 | igcewieling | tzafrir: it is possible to prevent a call from hanging up, but it usually requires some form of operator or 911 services. chan_dahdi has support for it. |
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| 15:11.56 | adnc | hello, I would like to test a peer from my asterisk box 1 to box 2 and for this I need to simulate a non reachable end. is there an easy way to disrupt that with a command on asterisk? |
| 15:14.03 | eric_hill | "core stop gracefully" would certainly make the box unreachable. |
| 15:14.20 | leifmadsen | +1 |
| 15:14.25 | leifmadsen | or block the port with iptables |
| 15:14.45 | leifmadsen | either on the source or the destination box |
| 15:14.59 | leifmadsen | you could iptables block it on the source machine so you don't need to actually make the remote machine unreachable |
| 15:15.14 | leifmadsen | then your automated script could remove the block after the test |
| 15:17.47 | adnc | i see |
| 15:21.18 | *** join/#asterisk module000 (~module000@ec2-54-225-236-107.compute-1.amazonaws.com) |
| 15:21.50 | adnc | leifmadsen, I don't know the iptables command by hard which would block traffic on iax2 port, so I used core stop gracefully which also stops asterisk remote |
| 15:22.08 | leifmadsen | learning iptables isn't too difficult |
| 15:22.21 | leifmadsen | you block a specific outbound port, not the whole IP |
| 15:22.24 | leifmadsen | but whatevs |
| 15:22.25 | leifmadsen | gl |
| 15:24.53 | adnc | I'm checking the peer status with asterisk -r -x "iax2 show peers" | grep $PEER | grep "OK (" |
| 15:25.16 | module000 | adnc: which port do you want to block? |
| 15:25.20 | adnc | with a daemon, but this adds lots of log entries in asterisk logs. is there a better way getting status |
| 15:25.25 | adnc | module000, iax2 |
| 15:25.34 | module000 | adnc: which literal port, i was going to type the iptables for you =P |
| 15:25.43 | adnc | ohh, 4569 |
| 15:26.04 | module000 | adnc: iptables -t filter -I OUTPUT -m udp -p udp --dport 4569 -j DROP |
| 15:26.13 | adnc | module000, thanks |
| 15:26.32 | adnc | module000, how do I delete this rule again? |
| 15:26.58 | module000 | to take that out, you'd type: iptables -t filter -D OUTPUT -m udp -p udp --dport 4569 -j DROP |
| 15:27.29 | adnc | module000, thank you |
| 15:27.34 | igcewieling | module000: you should have "that is left that as an exercise for the reader" |
| 15:28.06 | adnc | igcewieling, excercise for something that you only need once? |
| 15:28.09 | module000 | igcewieling: maybe, but i don't think that's too much handholding... ideally he/she will want to duplicate the effect, and will end up learning from the exercise? |
| 15:28.26 | igcewieling | module000: or come back here and ask |
| 15:28.34 | module000 | igcewieling: that's ok, then we get to abuse them? :) |
| 15:28.46 | igcewieling | adnc: you are running a PBX, if you are not using iptables you are not a good admin |
| 15:29.03 | adnc | igcewieling, administering the firewall is the job of someone else |
| 15:29.16 | igcewieling | If you are running FreePBX without iptables then you might as well just give hackers phone cards. |
| 15:29.20 | adnc | also that is a matter of investing time |
| 15:29.27 | adnc | but also that discussion here |
| 15:29.31 | adnc | with you |
| 15:29.43 | adnc | igcewieling, do you have phone cards? |
| 15:29.45 | adnc | I dont |
| 15:30.08 | adnc | module000, thank you very much again |
| 15:34.41 | *** join/#asterisk timholum (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 15:35.08 | timholum | Is there a way I can turn debuging on for just one channel? |
| 15:36.31 | [TK]D-Fender | sip/iax, yes. |
| 15:36.37 | [TK]D-Fender | Anything else? no |
| 15:37.37 | timholum | How would I set the debuging on just the one channel? |
| 15:38.41 | timholum | Nevermind |
| 15:38.58 | timholum | core set debug channel "channelid" I am assuming |
| 15:39.16 | timholum | I did not see the channel option in debug before |
| 15:40.22 | timholum | Thanks [TK]D-Fender |
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| 16:09.56 | Penguin | module000: If you would have specified port name 'iax' it would have worked without knowing the port number. |
| 16:11.05 | module000 | Penguin: if 'iax' is in /etc/services - then you're right. |
| 16:11.06 | *** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz) |
| 16:11.48 | Penguin | If it isn't in there, you need to update your box correctly. |
| 16:18.38 | *** join/#asterisk navaismo (~navaismo@189.241.92.175) |
| 16:21.51 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 16:21.59 | wasanzy | hi |
| 16:22.17 | WIMPy | lo |
| 16:22.56 | wasanzy | I have just finished installing asterisk, how do I know everything is working? any logs or debuging to show now error popups during start? |
| 16:23.32 | WIMPy | Did you read the |
| 16:23.36 | WIMPy | ~book |
| 16:23.37 | 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 |
| 16:24.27 | wasanzy | I read a little but not all yet |
| 16:25.03 | WIMPy | It should hekp ypu with your first steps. |
| 16:29.32 | [TK]D-Fender | wasanzy: You know it's working.. by using it. |
| 16:29.44 | wasanzy | ok |
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| 16:51.02 | anonymouz666 | Ohh, 4th edition was released. |
| 16:51.08 | anonymouz666 | I will buy |
| 16:51.10 | anonymouz666 | nice to know |
| 16:53.01 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
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| 16:58.53 | *** join/#asterisk CrashSys (~kumba@office2.vicidial.com) |
| 16:59.15 | CrashSys | Anyone know how to get the to: part of a sip invite from asterisk to be the domain not the host line of a sip entry? |
| 17:07.02 | igcewieling | the From domain? |
| 17:07.11 | CrashSys | the to: |
| 17:08.03 | igcewieling | check to see if the domain a record is the same as the hostname, if so you problem is solved |
| 17:08.12 | igcewieling | unlikely, but worth checking |
| 17:11.25 | WIMPy | CrashSys: Not possible. |
| 17:11.28 | CrashSys | host and domain are different |
| 17:12.22 | *** join/#asterisk mokmeister (~mokmeiste@86-40-146-176-dynamic.b-ras2.lmk.limerick.eircom.net) |
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| 18:20.19 | TheKernel-work2 | so I've installed Asterisk 11.5, I've made sure under make menuconfig that there is an * next to the res_rtp_asterisk module and its dependencies are all install, no errors during the make and mak install. But when I make a call I still get this error! ERROR[29337][C-00000002]: rtp_engine.c:259 ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? |
| 18:20.49 | TheKernel-work2 | I've searched like crazy and done everything I could and it still does this |
| 18:20.58 | Cuzner | libuuid? |
| 18:21.01 | TheKernel-work2 | not only that, but it does it on Fedora AND Ubuntu |
| 18:21.07 | TheKernel-work2 | Cuzner: got it installed |
| 18:21.19 | Cuzner | before you did the build of asterisk? |
| 18:21.33 | TheKernel-work2 | yes |
| 18:21.49 | Cuzner | sorry, i won't be of much help then. |
| 18:21.56 | TheKernel-work2 | dang |
| 18:22.27 | eric_hill | TheKernel-work2: apt-get install uuid, uuid-dev, libuuid1, and uuid-runtime |
| 18:22.46 | eric_hill | TheKernel-work2: Then ./configure --whatever |
| 18:23.03 | eric_hill | TheKernel-work2: Re-make && make install |
| 18:23.26 | eric_hill | I ran into that last week on a fresh clean 11.5 build. |
| 18:23.52 | TheKernel-work2 | I did not get uuid-runtime |
| 18:23.54 | Cuzner | right, you don't just need libuuid |
| 18:23.55 | TheKernel-work2 | let me try that |
| 18:23.58 | Cuzner | i think you need libuuid-devel as well |
| 18:24.19 | eric_hill | If uuid isn't installed when the initial configure happens, even installing it and re-making won't work. You have to re-configure so the presense of that library gets noticed. |
| 18:25.18 | eric_hill | Cuzner, I think you're right. FWIW, cat uuid config.log => UUID_LIB=' -luuid ' |
| 18:26.15 | TheKernel-work2 | ./configure is reconfiguring right? |
| 18:26.19 | TheKernel-work2 | re-configure* |
| 18:26.26 | eric_hill | Right. |
| 18:26.32 | TheKernel-work2 | eric_hill: looks like I had all that |
| 18:27.05 | TheKernel-work2 | I installed everything that had "uuid" in it onto my systems |
| 18:27.30 | eric_hill | Do you have res_rtp_asterisk.so in your modules directory? |
| 18:28.20 | eric_hill | And what does "module show like res_rtp" give you in the cli? |
| 18:28.51 | *** join/#asterisk jsjc (~Adium@103.Red-2-136-95.dynamicIP.rima-tde.net) |
| 18:28.54 | eric_hill | BRB |
| 18:29.42 | elguero | TheKernel-work: Are you installing from tarball? |
| 18:30.23 | TheKernel-work2 | elguero: yes |
| 18:30.23 | *** join/#asterisk navaismo (~navaismo@189.241.66.140) |
| 18:30.28 | TheKernel-work2 | I have to in order to get 11.5 |
| 18:30.38 | TheKernel-work2 | res_rtp_asterisk.so Asterisk RTP Stack 0 |
| 18:30.54 | elguero | nevermind then... saw somebody using patches to get up to 11.5 and had a problem |
| 18:36.45 | adnc | I'm checking the peer status with asterisk -r -x "iax2 show peers" | grep $PEER | grep "OK (" |
| 18:36.50 | adnc | with a daemon, but this adds lots of log entries in asterisk logs. is there a better way getting status |
| 18:37.00 | adnc | than the above command? |
| 18:38.27 | [TK]D-Fender | AMI |
| 18:38.44 | adnc | with manager interface? |
| 18:38.51 | [TK]D-Fender | That's what AMI stands for |
| 18:40.05 | eric_hill | TheKernel-work2: in sip.conf, do you have rtp_engine=asterisk |
| 18:43.07 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 18:45.33 | eric_hill | Does anyone run a production asterisk in Amazon's EC2 cloud? If so, any problems with jitter or IO bandwidth? |
| 18:45.41 | igcewieling | res_rtp_asterisk requires libuuid library and headers. If you don't have them, then Asterisk won't build. Once you add the required library, re-run ./configure and rebuild and reinstall. This is a FAQ |
| 18:45.52 | igcewieling | how do you add a factoid to the bot? |
| 18:46.23 | ChannelZ | infobot: my butt is huge |
| 18:46.23 | infobot | ...but my butt is already something else... |
| 18:46.32 | ChannelZ | kinda like that |
| 18:46.53 | eric_hill | Asterisk builds just fine without uuid, but just won't handle calls... |
| 18:47.00 | Cuzner | infobot: res_rtp_asterisk is res_rtp_asterisk requires libuuid library and headers. If you don't have them, then Asterisk won't build. Once you add the required library, re-run ./configure and rebuild and reinstall. This is a FAQ |
| 18:47.00 | infobot | Cuzner: what are you talking about? |
| 18:47.05 | Cuzner | :) |
| 18:47.26 | Cuzner | guess bot doesn't like me trying to add stuff |
| 18:48.06 | igcewieling | It is a really stupid design decision. |
| 18:49.58 | *** join/#asterisk mokmeister (~mokmeiste@86-40-146-176-dynamic.b-ras2.lmk.limerick.eircom.net) |
| 18:50.04 | navaismo | the infobot or asterisk uuid? |
| 18:50.22 | navaismo | seems like eric_hill can't catch the point yet |
| 18:52.03 | eric_hill | You said it won't build. It does build. Just doesn't work. Yes, that's moronic. If it can't handle calls, it shouldn't build. |
| 18:57.47 | igcewieling | no, res_rtp_asterisk will not build. |
| 18:58.02 | igcewieling | Asterisk will however, build and install just fine. |
| 18:58.33 | igcewieling | the OTHER issue with Asterisk 11 is you need to run an ldconfig after it initially installs libasteriskssl |
| 19:00.19 | TheKernel-work2 | eric_hill: yes we install 1.8 on amazon ec2 and it works great, very low jitter and the IO bandwidth is almost non-exsistant |
| 19:02.49 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.43) |
| 19:03.47 | ChannelZ | ~res_rtp_asterisk |
| 19:03.47 | infobot | i guess res_rtp_asterisk is "res_rtp_asterisk requires libuuid library and headers. If you don't have them, then Asterisk won't build. Once you add the required library, re-run ./configure and rebuild and reinstall." |
| 19:04.03 | ChannelZ | I got it to work with quoting, otherwise not sure why he is being difficult about it. |
| 19:05.46 | navaismo | <PROTECTED> |
| 19:06.39 | TheKernel-work2 | "Asterisk won't properly build" |
| 19:06.43 | TheKernel-work2 | would work I guess |
| 19:07.24 | eric_hill | Agreed. |
| 19:09.16 | eric_hill | BTW, thanks TheKernel-work2. I think an EC2 asterisk instance would make a great failover for our primary VM. |
| 19:13.09 | igcewieling | "res_rtp_asterisk won't build" |
| 19:15.12 | TheKernel-work2 | I fixed it |
| 19:15.19 | TheKernel-work2 | you have to remove the old modules |
| 19:15.43 | igcewieling | the modules listed when you do a make install 8-) |
| 19:15.44 | TheKernel-work2 | rm /usr/lib/asterisk/modules/res_rtp_asterisk |
| 19:16.06 | TheKernel-work2 | thne reconfigure and make make install |
| 19:16.19 | igcewieling | chances are you installed a previous version of Asterisk which did not need libuuid and is incompatible with whatever you are building now. |
| 19:16.41 | Aivaras | can someone share default codecs.conf file? |
| 19:17.03 | igcewieling | Aivaras: there is no default. |
| 19:17.24 | igcewieling | you might want to check the sample in your Asterisk source directrory under configs/ |
| 19:17.30 | Aivaras | if I delete everything in mine - it will be as default? |
| 19:19.36 | igcewieling | why not simply delete codecs.conf? |
| 19:19.58 | igcewieling | You only need it if you are running a few specific codecs like Speex |
| 19:21.25 | *** join/#asterisk colinwielga (~colinwiel@static-72-64-129-133.tampfl.fios.verizon.net) |
| 19:21.30 | *** part/#asterisk colinwielga (~colinwiel@static-72-64-129-133.tampfl.fios.verizon.net) |
| 19:21.46 | Aivaras | I changed settings of codecs for SIP in some web interface I don't have any more and I think now it's causing problems |
| 19:21.48 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 19:22.01 | Aivaras | Do I need to somehow reload codecs confing? |
| 19:22.16 | igcewieling | "module reload" should be enough. |
| 19:22.25 | igcewieling | I highly doubt it has anything do with your issues |
| 19:22.27 | *** join/#asterisk colinwielga (~colinwiel@static-72-64-129-133.tampfl.fios.verizon.net) |
| 19:23.46 | Aivaras | I disabled few codecs and now I don't know how to turn them on again |
| 19:24.57 | igcewieling | edit sip.conf and allow= them |
| 19:25.04 | igcewieling | also you should go read the Asterisk book |
| 19:25.06 | igcewieling | ~book |
| 19:25.06 | 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 |
| 19:30.30 | navaismo | ~buybook |
| 19:30.30 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/0636920025894 so go buy it SERIOUSLY |
| 19:30.44 | navaismo | ~buybookmx |
| 19:30.50 | navaismo | ~buybook-mx |
| 19:34.35 | colinwielga | ~book |
| 19:34.36 | 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 |
| 19:34.49 | colinwielga | ~buybook |
| 19:34.49 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/0636920025894 so go buy it SERIOUSLY |
| 19:35.48 | Cuzner | alright, how much is leifmadsen paying you guys to spam those macros? :P |
| 19:35.57 | *** join/#asterisk Busstech (~Busstech@41-133-18-37.dsl.mweb.co.za) |
| 19:36.05 | Busstech | Good day all |
| 19:36.41 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:36.42 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:38.15 | navaismo | Cuzner, I can't talk about it |
| 19:38.40 | navaismo | he force me to sign an evil NDA |
| 19:38.57 | Cuzner | heh |
| 19:40.25 | Qwell | Cuzner: For every extra book he sells, he doesn't murder us. |
| 19:43.27 | Busstech | Using Asterisk 8.4.x how do I change the dial plan so the cdr records outgoing numbers? |
| 19:43.48 | Cuzner | Qwell: well, i guess you could call that an incentive... |
| 19:47.06 | *** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 19:48.10 | igcewieling | Busstech: Start by providing a valid Asterisk version |
| 19:50.00 | Busstech | my bad, Asterisk 1.8.23 |
| 19:56.16 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 19:57.16 | navaismo | Busstech, the cdr records all calls(maybe the not the no-answered until you enable it), what is your issue? |
| 19:58.56 | Busstech | Basically what happens is the car write to MySQL but the number is only written to the row in the event that the caller doesn't pick up the phone. if the called picks up the phone, the record in the DB doesn't reflect the number dialed. |
| 19:59.57 | Busstech | Sorry if I can't answer all the questions 100% Im trying to find a solution to what my developers are able to explain to me. |
| 20:02.01 | navaismo | so you are using an external program to write the cdr? |
| 20:03.14 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 20:03.42 | pabelanger | anybody know if voip.ms will support speex? |
| 20:04.28 | Busstech | hmm, not as far as I understand. asterisk writes the cdr |
| 20:06.27 | navaismo | Busstech, i cant understand your issue |
| 20:06.56 | navaismo | maybe if you provide an output for the cli & explain the error |
| 20:10.46 | [TK]D-Fender | pabelanger: http://voip.ms/faq.php#supportedcodecs |
| 20:10.56 | [TK]D-Fender | pabelanger: They hid it .... in the big print :) |
| 20:11.40 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:11.40 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:17.29 | *** join/#asterisk jkroon (~jkroon@41.16.92.254) |
| 20:18.36 | jkroon | WIMPy, Qwell, Greenlight - just figured I'd give some feedback, it's now been >24h after I moved the astdb onto tmpfs, and not a single SIP request took >1ms since. Compared to >150ms previously for REGISTER requests, even when not under load. |
| 20:18.57 | navaismo | Busstech, the dst of the cdr store the dialed number |
| 20:19.31 | navaismo | s/dst/dst field/ |
| 20:20.23 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.236) |
| 20:20.27 | *** join/#asterisk thegoat (~thegoat@c-68-32-48-69.hsd1.pa.comcast.net) |
| 20:20.35 | jkroon | navaismo, not quite true |
| 20:20.56 | jkroon | not if you rewrite it, it contains ${EXTEN} that was last used. |
| 20:21.09 | thegoat | i am in the process of doing a fresh install of asterisk, and what would the differences in using the built in skinny vs chan_sccp_b? |
| 20:23.09 | Busstech | Thanks folks, think it would be best if I get my developers to chat here. our firewall policy is really bad so I'll take my lappy to work tomorrow for them to use. Thanks for the guidance. |
| 20:25.38 | navaismo | jkroon, talking about normal behavior |
| 20:43.51 | navaismo | So i followed this in order to install pjproject https://wiki.asterisk.org/wiki/display/AST/Installing+pjproject but when i run the cmd ldconfig -p | grep pj the modules are installed in /lib, even if I set the prefix in configure and Asterisk cant detect the modules and can't compile the res_sip resource |
| 20:50.52 | Qwell | navaismo: hold that though |
| 20:51.55 | Qwell | +t |
| 20:52.29 | Qwell | navaismo: git pull, ./configure again, make, make install |
| 20:52.52 | navaismo | ok |
| 20:58.55 | *** join/#asterisk sidus (~abracadab@37-5-73-205-dynip.superkabel.de) |
| 21:01.24 | navaismo | taking a looooong time to compile |
| 21:02.29 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:03.39 | *** join/#asterisk eject_ck (~Eugene@62.205.134.210) |
| 21:07.23 | *** join/#asterisk felipealmeida (~user@177.205.229.186.dynamic.adsl.gvt.net.br) |
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| 21:09.32 | leifmadsen | I think all asterisk versions should go back in history and have the leading 1 stripped |
| 21:09.34 | *** join/#asterisk slidesinger (~slidesing@c-69-141-78-33.hsd1.nj.comcast.net) |
| 21:09.40 | leifmadsen | I shall forever refer to asterisk 1.4 as asterisk 4 |
| 21:09.41 | leifmadsen | :D |
| 21:10.51 | newtonr | leifmadsen: Nah we have used numbers long enough. We need to switch to letters now. |
| 21:11.08 | leifmadsen | newtonr: ABE C.2.5.4 |
| 21:11.12 | leifmadsen | been there done that |
| 21:11.22 | newtonr | shhh |
| 21:11.28 | leifmadsen | you shut your mouth when you're talking to me |
| 21:11.32 | leifmadsen | :D |
| 21:12.07 | navaismo | O_O |
| 21:12.15 | newtonr | leifmadsen: lol |
| 21:14.29 | Qwell | leifmadsen: Nobody uses C.2 anymore. |
| 21:14.32 | Qwell | get with it |
| 21:20.12 | leifmadsen | Qwell: oh don't i know it |
| 21:20.21 | leifmadsen | Qwell: actually I likely have a former client running a custom C.x |
| 21:38.26 | navaismo | Qwell, after that asterisk can see the modules, thanks a lot. |
| 21:39.20 | *** join/#asterisk smkelly (~smkelly@mykonos.smkelly.org) |
| 21:41.15 | smkelly | So, lets say I'm about to embark on a project to redo all of my Asterisk configs in a way that doesn't suck. Lets also say I'm not a fan of the extensions.conf format. Would I be insane or regret it down the road if I redid it all in AEL or Lua? |
| 21:41.31 | smkelly | glares at file |
| 21:43.46 | Kobaz | ael +1 |
| 21:44.06 | smkelly | this file guy told me that AEL isn't supported, or is supported less, or something about how I'm crazy |
| 21:44.15 | igcewieling | smkelly: We have a small amount of extensions.conf and call AEL macros for virtually everything using AELsub |
| 21:44.16 | Kobaz | it works |
| 21:44.19 | smkelly | but he's the crazy one, so i figured I'd come ask real people |
| 21:44.20 | Kobaz | it doesnt need to be supported really |
| 21:44.29 | Kobaz | there are some things here and there that could be better |
| 21:44.34 | Kobaz | which i was actually going to take care of |
| 21:44.41 | igcewieling | smkelly: file is one of the Old Ones, but not everyone agrees AEL should not be used. |
| 21:44.44 | Kobaz | but it works, and it's 100x better than extensions.conf |
| 21:45.01 | Kobaz | and if you're going to think about doing lua, just use agi instead and fire up your favorite language |
| 21:45.04 | smkelly | file: in yo' and @MsZoeDog's face |
| 21:45.37 | smkelly | Lua was kind of not high on the list |
| 21:46.01 | [TK]D-Fender | There is nothing you can do in AEL you can't do in standard. The reverse is not true. |
| 21:46.25 | [TK]D-Fender | AEL is an extra parsing layer and by the time it's done parsing your code back it makes debugging much more difficult |
| 21:46.26 | Kobaz | anything in extensions.conf you can do in ael |
| 21:46.28 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 21:46.30 | Kobaz | and vise versa |
| 21:46.46 | Kobaz | well, logic-wise |
| 21:47.00 | Kobaz | niceness-wise, there's many things in ael you cannot do in extensions.conf |
| 21:47.11 | [TK]D-Fender | Kobaz: Like? |
| 21:47.19 | Kobaz | [TK]D-Fender: it's not that it's harder to debug, it's just that you need to debug differently |
| 21:47.24 | Kobaz | [TK]D-Fender: structured scripting? |
| 21:47.27 | [TK]D-Fender | Kobaz: Since it gets parsed back... I can't think of any possibility of ANY |
| 21:47.33 | [TK]D-Fender | that is style, not capability |
| 21:47.38 | Kobaz | code blocks? if/then, case statements? |
| 21:47.46 | [TK]D-Fender | Functioannly all parsed back. |
| 21:47.53 | Kobaz | the style is a capability |
| 21:47.59 | smkelly | I mean sure you coudl do it in extensions.conf, it'd be like doing jump tables in assembly |
| 21:48.00 | Kobaz | you can change your style in extensions.conf |
| 21:48.12 | Kobaz | that still doesn't mean that it's as easy to work with or pretty to look at |
| 21:48.22 | [TK]D-Fender | Kobaz: Can you make a context with ONLY a priority 2 for a given exten using AEEL? |
| 21:48.23 | igcewieling | Kobaz: anything you can do in AEL you can do in extensions.conf if you are willing to tie yourself into knots doing it. |
| 21:49.00 | igcewieling | If you can't do something in AEL then use extensions.conf for that. |
| 21:49.18 | Kobaz | [TK]D-Fender: well okay, in some sort of contrived example there's some things... but you could do exten => { NoOp(); DoSomethingUseful(); } |
| 21:49.22 | Kobaz | there you go, there's your priority 2 |
| 21:49.37 | Kobaz | but why would you want to even *think* about a priority other than 1, if you're going to use ael in the first place |
| 21:49.56 | [TK]D-Fender | Kobaz: because of use of INCLUDE order to customize dialplan functionality |
| 21:50.12 | *** join/#asterisk mduell (~mduell@natpool.gwp.corp.flightaware.com) |
| 21:50.19 | Kobaz | context a { includes { foo; bar; baz; } } |
| 21:50.22 | Kobaz | there's your includes, in order |
| 21:50.32 | [TK]D-Fender | No. |
| 21:50.35 | Kobaz | yeap |
| 21:50.38 | [TK]D-Fender | I'm talking about allowing OVERLAP |
| 21:50.40 | [TK]D-Fender | ^ |
| 21:50.47 | [TK]D-Fender | single priorities |
| 21:50.56 | Kobaz | you can overlap |
| 21:51.06 | Kobaz | do your context, define some extensions, include some stuff |
| 21:51.17 | [TK]D-Fender | but I don't WANT priority 1 overridden |
| 21:51.23 | [TK]D-Fender | just #2 |
| 21:51.27 | Kobaz | so then don't override it |
| 21:51.32 | igcewieling | Kobaz: [TK]D-Fender is one of those people who think "exten => _1NXXNXXXXXX,1,Whatever" with "exten => 12125551212,2,whatever" without a priority 1 is not an abomination against nature. I happen to disagree. |
| 21:51.38 | [TK]D-Fender | But I can't just have priority 2 |
| 21:51.45 | Kobaz | [TK]D-Fender: then don't design it like that |
| 21:51.51 | Kobaz | use if/then/else |
| 21:51.59 | [TK]D-Fender | How Apple of you... |
| 21:52.03 | Kobaz | logic-wise there's nothing you can't do |
| 21:52.13 | igcewieling | Kobaz: you'll never convince [TK]D-Fender. |
| 21:52.14 | Kobaz | if you want to compare exact-functional wise, i agree |
| 21:52.18 | [TK]D-Fender | It's not that "you can't do it", it's that "you shouldn't WANT it" |
| 21:52.22 | Kobaz | but you don't need a 1-to-1 |
| 21:52.36 | [TK]D-Fender | Same thinking on my iOS can't use WEP keys otehr than 1 |
| 21:52.41 | igcewieling | [TK]D-Fender: in my opinion you should not WANT to do it the way you are describing way. |
| 21:52.54 | Kobaz | it's like complaining i can't very easily swing this chisel trying to split wood |
| 21:52.57 | Kobaz | versus using an axe |
| 21:53.03 | Kobaz | they accomplish the same thing, split wood |
| 21:53.06 | [TK]D-Fender | igcewieling: that's just one sample. You know I'll find more... and more practical ones as I go. |
| 21:53.10 | Kobaz | but you have to use your head to think differently about how to use it |
| 21:53.31 | Kobaz | you use a hammer on the chisel, but it gets the job done |
| 21:53.42 | Kobaz | using an axe is far superior |
| 21:53.47 | Kobaz | ie: ael :P |
| 21:54.01 | Kobaz | you'll get more fine grained control with the hammer and chisel |
| 21:54.07 | Kobaz | it'll also take you 10 times longer do to your job |
| 21:54.38 | [TK]D-Fender | And when you go to the next level of "centralized" and "smart" management.. you'll want it in a DB. At which point kiss AEL goodbye |
| 21:54.45 | Kobaz | yeah |
| 21:54.47 | Kobaz | but you need glue |
| 21:55.03 | [TK]D-Fender | And nobody needs AEL |
| 21:55.05 | Kobaz | your call is going to hit dialplan sooner or later |
| 21:55.12 | Kobaz | might as well make it easy to maintain, and pretty to look at |
| 21:55.20 | [TK]D-Fender | its one more layer and only slows down execution and introduces another point of failure |
| 21:55.41 | [TK]D-Fender | Maintaim... maybe. debug ,no. |
| 21:55.41 | Kobaz | in 5 years of using ael, i've never had a "failure" |
| 21:56.06 | Kobaz | logic issues, yes, but that's fixable with a text editor and "ael reload" |
| 21:56.56 | smkelly | the db argument is valid I suppose |
| 21:57.16 | smkelly | now if AEL had funcs to make ODBC calls.. |
| 21:57.47 | Kobaz | from my db, i generate one line, maybe two lines of extensions.conf extensions, which go to ael, which does some small decision making and then fires off agi |
| 21:58.05 | Kobaz | it's super flexible, really easy to maintain, super easy to train someone new, and headache free |
| 21:58.17 | smkelly | so you use both .conf and .ael? |
| 21:58.33 | Kobaz | your forced to use .conf if you want to load extensions.conf from a db |
| 21:58.42 | Kobaz | or do db generated dialplan in any kind of sane way |
| 21:58.55 | Kobaz | you could generate ael from the db, but for a one line goto.. that's silly |
| 21:59.28 | smkelly | So do the two get merged? |
| 21:59.30 | Kobaz | like, i have a table of sip extensions |
| 21:59.31 | Kobaz | yeah |
| 21:59.39 | Kobaz | so i have a view, that makes extensions.conf |
| 21:59.56 | Kobaz | and i have like.. context myphones... 1000, 1001, 1002.. all db generated |
| 22:00.04 | Kobaz | and they all do a goto dialExten(${EXTEN}); |
| 22:00.05 | Kobaz | and that's it |
| 22:00.08 | *** join/#asterisk sebastianpersic (~chatzilla@ua-85-227-32-4.cust.bredbandsbolaget.se) |
| 22:00.19 | Kobaz | dialExten lives in ael, does all the initial work |
| 22:00.27 | Kobaz | and then calls AGI |
| 22:00.38 | smkelly | What do you do with AGI? |
| 22:00.38 | *** join/#asterisk sebastianpersic (~chatzilla@ua-85-227-32-4.cust.bredbandsbolaget.se) |
| 22:00.39 | *** join/#asterisk andrewyager (~andrewyag@120.159.206.19) |
| 22:01.01 | Kobaz | agi does call routing |
| 22:01.11 | Kobaz | figures out which trunk group you want to go out |
| 22:01.16 | Kobaz | finds the first trunk, checks the capacity |
| 22:01.23 | Kobaz | does any number mangling |
| 22:01.28 | Kobaz | like remove 9, add 1 |
| 22:01.29 | Kobaz | sort of thing |
| 22:01.34 | *** join/#asterisk andrewya_ (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 22:01.50 | Kobaz | the ael basically sets up some variables, and tells the agi what phone is calling what number |
| 22:02.20 | Kobaz | like, some things you don't need agi for |
| 22:02.23 | Kobaz | like my voicemail handler |
| 22:02.24 | WIMPy | Oh, damn, I missed jkroon. |
| 22:03.40 | Kobaz | http://pastebin.com/pC754UGa |
| 22:03.41 | Kobaz | so |
| 22:03.52 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 22:04.00 | Kobaz | tell me that's not 100x easier to look at than extensions.conf spaghetti |
| 22:04.50 | smkelly | sooo better |
| 22:08.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:12.02 | *** part/#asterisk sebastianpersic (~chatzilla@ua-85-227-32-4.cust.bredbandsbolaget.se) |
| 22:14.04 | *** join/#asterisk guitarHester (~guitarHes@nat/digium/x-oauorerzbcxyvlhn) |
| 22:15.00 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 22:20.37 | revolve | Kobaz: what language is that? |
| 22:22.27 | navaismo | The column Status for the output of a module show... its very useful. You rock guys! |
| 22:23.26 | robl^ | Kobaz: you should look at lua ;-) I used to use AEL2 then found that I could write my dialplan in lua |
| 22:24.12 | smkelly | robl^: what do you do in lua that is easier? |
| 22:24.53 | navaismo | The sip.conf and res_sip complement each other or are different stuff? |
| 22:25.02 | *** join/#asterisk bbs (~bbs@bbs71364-sbx.creighton.edu) |
| 22:26.00 | robl^ | smkelly: I do complex IVRs. IT's a bit easier to deal with lots of if/then clauses, string comparisons, etc with lua. at least it was for me |
| 22:27.15 | robl^ | my extensions.conf has 5 lines. I have more than 5,000 lines in lua broken up into about 20 files |
| 22:29.50 | navaismo | in the wiki exist the documentation for the res_sip.conf? |
| 22:31.24 | ChannelZ | I don't even know what res_sip.conf is |
| 22:32.09 | robl^ | smkelly: it's not that I couldn't do it in another language, i.e. the traditional dialplan extensions.conf.. but I chose one that was easier for me to manage in the long term |
| 22:33.11 | ChannelZ | Oh. Asterisk 12 stuff. |
| 22:37.01 | WIMPy | res_sip must be part of chan_pjsip. |
| 22:37.37 | *** join/#asterisk guitarHester (~guitarHes@nat/digium/x-uurapfuattewfjro) |
| 22:41.29 | *** join/#asterisk eject_ck (~Eugene@62.205.134.210) |
| 22:41.30 | elguero | WIMPy: ya, I am waiting for jkroon to get online again too... his findings with astdb intrigued me and I did some research on sqlite3 and came up with a patch for astdb that seems to result in a slight performance improvement (especially with updates)... not sure if it is enough to fix the issue he was happening but it is an improvement based on a very, very simple test I did.. need someone with a lot |
| 22:41.37 | elguero | of registrations or better test setup to test the patch and report back |
| 22:41.58 | *** join/#asterisk zeroschism (ajs07635@147.134.4.74) |
| 22:42.51 | WIMPy | elguero: I also did some tests yesterday and found out a lot of things I didn't really want to know. |
| 22:42.58 | WIMPy | The story is here: http://wimpy.yeti.dk/pastebin |
| 22:43.11 | elguero | WIMPy: yep, I saw that when you posted it in -dev |
| 22:43.20 | WIMPy | ok |
| 22:43.26 | elguero | it is quite interesting... hopefully we can figure out what changed |
| 22:43.41 | WIMPy | At least. |
| 22:44.22 | WIMPy | The stating point for me was when I realized that it took 2.4 to 3 seconds for any phone to ring after a call came in. |
| 22:44.36 | WIMPy | With only 20 lines of dialplan, that is. |
| 22:45.12 | WIMPy | So the current one is hardly usable. |
| 22:45.51 | WIMPy | But I find the media thing rather shocking as well, especially what file just said. |
| 22:46.14 | file | it's been like that for, well, ever |
| 22:46.31 | WIMPy | That doesn't make it better. |
| 22:46.46 | file | sure, but changing that has far reaching consequences |
| 22:46.56 | WIMPy | Like I just said in -dev it could always happen to anyone. |
| 22:47.17 | file | and I think without REALLY looking into things deeply just assuming that stuff is the reason is not useful |
| 22:47.31 | WIMPy | And curently I'm seeing, or rather hearing, it witrh a simple Dial(). |
| 22:47.39 | file | and I'm not |
| 22:48.12 | elguero | I was going to say, I am running latest trunk on home phone system and not getting that problem either |
| 22:48.29 | WIMPy | That is when I answer a call, I better wait for a second or two before saying anything as it will get lost otherwise. |
| 22:49.02 | *** join/#asterisk felimwhiteley_ (~quassel@89.101.203.26) |
| 22:49.07 | WIMPy | elguero: I'm sure it is amplified by some magnitude by the loopback switches I have. |
| 22:49.29 | WIMPy | Although I don't know what its doing in the context containing them. |
| 22:49.44 | WIMPy | The current call is not there. |
| 22:50.20 | WIMPy | Oh, and the impact of the loopback switches is then further amplified by DUNDi. |
| 22:50.48 | mic_ | hello, escaping operators => \+ <- that is ok, right? |
| 22:51.10 | WIMPy | mic_: Escape where? |
| 22:51.21 | file | yes, put it all together and you have an uncommon complicated setup which exercises many different code paths |
| 22:51.57 | mic_ | WIMPy: I wish to detect +45 sequence |
| 22:51.58 | WIMPy | Yes, but unfortunatly they are neccessary. |
| 22:52.02 | mic_ | WIMPy: in the dialplan - inside an IF |
| 22:57.47 | mic_ | ok, found an issue |
| 22:57.49 | mic_ | :) |
| 22:57.51 | mic_ | now it works. |
| 22:58.08 | WIMPy | mic_: I hoped someone else would answer that. I didn;t run into that issue, yet, but I can imagine quite interesting effects. |
| 22:58.44 | mic_ | WIMPy: yes, the key was to make sure you tell asterisk to make a comparison as _strings_ |
| 22:59.02 | mic_ | i.e. "${SIP_HEADER(To):5:3}" = "+45" |
| 22:59.26 | mic_ | and then it worked out immediately. |
| 22:59.47 | WIMPy | Yes, I guess, in that casw quotes do what you expect. |
| 23:00.23 | [TK]D-Fender | Quotes are literal.. there is no real "type" |
| 23:00.53 | mic_ | Thanks for clarification :) |
| 23:00.55 | WIMPy | Yes, but I can see how quotes force the type. |
| 23:01.09 | [TK]D-Fender | I don't since there is no storage difference. |
| 23:01.27 | [TK]D-Fender | I'm jsut not sure how it "escapes" the first + in the function call... |
| 23:02.24 | WIMPy | I think Asterisk tries to treat all values as numeric if possible. If you quote them it isn't. |
| 23:02.58 | mic_ | that's what I found out, too |
| 23:03.11 | mic_ | I would assume that the following would work properly |
| 23:03.22 | mic_ | ${SIP_HEADER(To):5:1} = 4 |
| 23:03.41 | [TK]D-Fender | makes you wonder what it thinks when there is clearly text on the right side of the operator then and then why it takes the quotes as literal then. |
| 23:03.52 | WIMPy | (isn't possible, that is) |
| 23:06.59 | mic_ | I am out of juice. Time to hit the bed. |
| 23:07.48 | mic_ | At least one problem from the "Ministry of Silly Comparisons" was solved tonight ;) |
| 23:08.24 | WIMPy | would like to see a solved problem as well. |
| 23:08.39 | WIMPy | But it looks like I just found the next one :-( |
| 23:08.49 | mic_ | I have a cool one for tomorrow |
| 23:09.06 | mic_ | I mean... a cool one :S |
| 23:30.14 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 23:30.14 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 23:35.35 | *** join/#asterisk artyx (U2FsdGVkX1@schizophrenia.googleplex.net) |
| 23:35.44 | artyx | Has anyone here used the Cisco IP phones with their asterisk server? |
| 23:36.01 | artyx | Specifically im conceiving of screwing aroudn with the cisco IP-7940 |
| 23:36.17 | artyx | Not can it be done, does anyone HERE do it |
| 23:44.44 | slav3_kitten | i have |
| 23:44.53 | slav3_kitten | don't do it artyx |
| 23:45.10 | slav3_kitten | save yourself the frustration, headache, and sleepless nights by going polycom |
| 23:45.43 | slav3_kitten | if i didn't already have 8 assorted 7960/7940/7911 phones i'd have gone polycom |
| 23:45.54 | slav3_kitten | expected ZERO support from cisco |
| 23:46.09 | slav3_kitten | expect online tutorials to be inaccurate |
| 23:46.19 | slav3_kitten | expected=expect |
| 23:46.35 | slav3_kitten | expect firmware to be iffy at best |
| 23:48.01 | slav3_kitten | on the other hand cisco inline power switches are cheap, cisco 7900 series phones are cheap, they do work well when you manage to get them working, however i don't recommend spending the time and effort on them professionally at all |
| 23:48.03 | artyx | we've already went with the 7940's and are using UCM |
| 23:48.07 | artyx | im so friggin tired of the UCM |
| 23:48.09 | navaismo | So using asterisk 12 trying to register a device using the res_sip.conf but it failed, this is my res_sip.conf and the output of the sip debug.--->http://pastebin.com/XCNf5iCx |
| 23:48.39 | slav3_kitten | artyx, you can get them to work decently with asterisk |
| 23:48.44 | slav3_kitten | but it's an uphill battle |
| 23:48.48 | artyx | There are tftp example files out there, so how hard can it be..... ITs not like theres a ton of extra features on these |
| 23:48.56 | artyx | Compare to yealink ? |
| 23:49.01 | artyx | Easier, harder, etc |
| 23:49.19 | artyx | this is all assuming we convert them to sip, as opposed ot skinny/sccp |
| 23:50.05 | slav3_kitten | yea. i've zero experience with the skinny/sccp but was told to stay away from it by people who had done skinny/sccp asterisk with them |
| 23:50.13 | artyx | gotcha |
| 23:50.28 | artyx | well i want to do it anyways if nothing else ot learn why not to do it :P |
| 23:50.36 | artyx | i just wont happen to do it with everyones phone at the same time |
| 23:50.41 | slav3_kitten | good luck :) |
| 23:50.45 | artyx | Maybe just 4 or 5 phones and test it out for awhile :P |
| 23:50.55 | artyx | I despise these phones tbh |
| 23:51.05 | slav3_kitten | once you get it hammered out on 1 phone, it scales well to others |
| 23:51.09 | artyx | Ya |
| 23:51.19 | artyx | There are other issues when scaling from 4 phones to 400 |
| 23:51.48 | slav3_kitten | like /accidentally/ burning down the building so you end up with polycom phones |
| 23:52.24 | slav3_kitten | i love my cisco phones, just not the uphill battle to get them to do what i wanted |
| 23:52.45 | slav3_kitten | still can't get the ntp clocks working on them for some fucking reason |
| 23:53.20 | navaismo | I need to ask here or in dev channel? |
| 23:53.44 | artyx | hmmmm.. i know with my sip yealinks, i had time issues until i told htem to use dhcp time |
| 23:53.49 | artyx | wonder if theres an option like that for cisco |
| 23:55.49 | slav3_kitten | if there is, i can't find it. and cisco refuses to tell you everything when using phones in sip mode |
| 23:56.04 | slav3_kitten | iirc their sip firmware is an epic kludge and not really supported |
