| 00:20.25 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:20.25 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 06:25.28 | goga | Hello. Need help with voicemail.conf in asterisk. TODO: send messages about voicemail to telegram channel. Problem: added mailcmd with custom script, but it not passing variables like ${VM_DUR} to script. Any help? |
| 06:29.43 | goga | Asterisk 13.15.0 FreeBSD 11.0-RELEASE-p1 |
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| 07:54.36 | goga | Hi again. I've lost connection with freenode. Did I missed some answers about args for mailcmd in voicemail.conf? |
| 07:55.49 | wdoekes | nope |
| 07:56.33 | lorsungcu | goga: mailcmd doesnt get those variables, afaik |
| 07:56.55 | lorsungcu | what i do is compose the email with those variables exposed somehow, then parse it out in a mailcmd script |
| 07:57.04 | lorsungcu | if mailcmd does expose them, i'd love to know about ti |
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| 07:57.35 | goga | So i have to deliver mail to local box and parse it? |
| 07:57.57 | lorsungcu | you could do that, i just pass the entire message into my script and do it there |
| 07:58.43 | goga | Thank you for your advice. |
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| 08:03.16 | Onyx47 | hello, does anyone know if there's a syntax that will make the new autohints functionality work when using AEL for dialplans? The parser doesn't like me just putting autohints=yes in there... |
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| 10:45.40 | pawiecki | Hi, I know it's maybe not 100% on topic, but I need some tips with DECT. I have this old KWS6000 system, with ~60 Base stations and ~90 Handsets. It wasn't deployed in a proper way, and my job now is to fix some minor errors. Right now I'm looking to re-do the sync chain, to have it's source in the middle of the system and better sync tree. It's a multi-level building, and I'm not sure how to plan it - make it floor-by-floor or jump-around-the-best-sig |
| 10:45.42 | pawiecki | nal-strength chain type? If you have some experience with this sort of thing, any advice would be appreciated. |
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| 11:55.19 | dunderproto | What are some benefits of running your own sip server as opposed to getting free accounts? |
| 11:56.08 | Samot | A lot. |
| 11:57.06 | dunderproto | Samot: Would you be able to point me to a website that explains the benefits? I'm considering setting up a personal server for conference calls of ~1-5 people |
| 11:57.11 | dunderproto | ~2-5* |
| 11:58.14 | Samot | Asterisk |
| 11:58.30 | Samot | If you just want to use it for conference services, just use Asterisk. |
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| 12:16.28 | dunderproto | Samot: Might asterisk be excessive for just personal use? |
| 12:20.52 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:23.07 | [TK]D-Fender | dunderproto, excessive is up to you |
| 12:24.31 | [TK]D-Fender | <dunderproto> What are some benefits of running your own sip server as opposed to getting free accounts? <- this question doesn't work really |
| 12:25.19 | [TK]D-Fender | dunderproto, * is a PBX & telephony toolkit. A "free account" is a service of some kind. * processes calls, it is not itself a "service" |
| 12:27.34 | dunderproto | [TK]D-Fender: true, I understand your point. I guess I mean, would it be worthwhile to learn how to set up asterisk myself...but perhaps this is a question I have to answer myself |
| 12:29.52 | [TK]D-Fender | dunderproto, indeed. describe your initial goal more clearly |
| 12:32.23 | dunderproto | [TK]D-Fender: I currently have a sip provider that lets me place public telephone calls, which I use everyday for work. I connect with a softphone on my laptop. However, recently I needed to get sip accounts for at least 2 people for group calling. So, I need to create at least 2 SIP accounts for them. I have no idea how telephony works, as you can probably perceive from my sometimes illogical |
| 12:32.29 | dunderproto | questions. |
| 12:34.13 | dunderproto | I'm wondering if I should 1) search for a free sip provider, like linphone or ostel, or 2) set up my own sip server. I'm leaning towards the second because it sounds like fun |
| 12:35.02 | [TK]D-Fender | last I heard linphone was a softphone, not a provider. |
| 12:35.21 | dunderproto | [TK]D-Fender: they are indeed a softphone, but they also offer free accounts I think |
| 12:35.49 | [TK]D-Fender | Your #2 doesn't make sense. Again, Asterisk does not give you service |
| 12:36.39 | dunderproto | [TK]D-Fender: Ah. |
| 12:36.53 | [TK]D-Fender | Like comparing a cellular phone company's PLAN ... to a cell PHONE. |
| 12:37.07 | [TK]D-Fender | * != service |
| 12:37.18 | [TK]D-Fender | * can be like an answering machine FOR your service |
| 12:37.34 | [TK]D-Fender | You still have to have service from somewhere to contact the real world |
| 12:38.02 | dunderproto | I just need from sip account to sip account, so I don't need the public telephone network |
| 12:38.32 | [TK]D-Fender | Think of what companies use as a "phone system". Menus for callers, voicemail, corporate directories, multiple phones as inside "extensions", conferencing, |
| 12:39.02 | Samot | dunderproto: This sounds more like you need a conference call services more than a PBX |
| 12:39.10 | [TK]D-Fender | So just for you to talk free amongst a few people you will give access to? |
| 12:39.35 | [TK]D-Fender | And they'll use a SIP device of their own that you'd give them connection info for? |
| 12:39.37 | dunderproto | [TK]D-Fender: Yes, I just need a way to call a@domain.com, b@domain.com, and c@domain.com simultaneously |
| 12:39.47 | dunderproto | they'll use their smartphones running linphone or csipsimple |
| 12:39.50 | [TK]D-Fender | Ok, * can do that |
| 12:39.56 | Samot | Yup. |
| 12:39.57 | dunderproto | Samot: Yes, I think perhaps I am not in the right place |
| 12:40.08 | Samot | dunderproto; Asterisk can do this |
| 12:40.21 | Samot | The real question is, what you need to do with the conf. calls.... |
| 12:40.27 | Samot | Is it for fun or is it business? |
| 12:40.35 | dunderproto | business, so reliability is important |
| 12:40.42 | Samot | Then in this case.. |
| 12:40.49 | Samot | I'd suggest paying for a service. |
| 12:40.56 | Samot | While installing Asterisk, learning it.... |
| 12:41.05 | [TK]D-Fender | If you want to get off the ground fast and think you might want to learn, but maybe slower and later you could start with a distro that installs a fully ready installation of * + a GUI to manage the basics |
| 12:41.08 | Samot | And then moving to it when you are comfortable with it and understand it. |
| 12:41.37 | Samot | Jumping straight into it to solve a business need with ZERO knowledge or experience...bad. |
| 12:41.45 | Samot | It will be a bigger PITA for you than you think. |
| 12:41.55 | Samot | If you need a solution now, pay for a service. |
| 12:41.59 | Samot | But install Asterisk. |
| 12:42.00 | Samot | Learn it |
| 12:42.11 | Samot | Get it working to a point you understand it and can deal with it.. |
| 12:42.15 | Samot | Then use it. |
| 12:42.23 | [TK]D-Fender | freepbx.org <- get their latest stable distro and install it and it cen get your basics without having to learn the heavy parts right away. |
| 12:42.28 | Samot | Yes. |
| 12:42.34 | dunderproto | [TK]D-Fender, Samot: Thanks for your advice. That really helps. I will do some research |
| 12:42.36 | Samot | That would be the way to go to learn. |
| 12:42.58 | [TK]D-Fender | And then as you're ready you can extend it with a bit of your own custom call processing, and decide if you want a completely DIY setup after |
| 12:43.13 | Samot | Yup. |
| 12:43.16 | Samot | Agreed |
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| 12:59.27 | dunderproto | Might Kamailio be better suited for my needs? |
| 12:59.36 | Samot | No |
| 12:59.43 | Samot | It doesn't do media. |
| 13:00.09 | dunderproto | You mean, it doesn't handle video? Or it doesn't handle voice? I only need voice, not video |
| 13:01.15 | Samot | It doesn't handle media |
| 13:01.30 | Samot | Media = audio and video |
| 13:01.37 | dunderproto | ah, I see |
| 13:01.42 | dunderproto | Samot: Thanks |
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| 13:04.15 | [TK]D-Fender | FreePBX is your best starting point |
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| 13:42.13 | klaxa|work | hi, this may be quite the newbie question: when i run asterisk -x "core show channels concise" it shows two connections if there is one call ongoing and i've been wondering for a while what the difference between "Dial" and "AppDial" is |
| 13:42.25 | klaxa|work | and i can't really find stuff with google |
| 13:42.44 | klaxa|work | is there like a glossary of terms used in asterisk? i feel very lost trying to read the docs |
| 13:42.53 | klaxa|work | i'm not good with phones |
| 13:48.31 | [TK]D-Fender | Each leg is a channel |
| 13:48.54 | [TK]D-Fender | Dial() is an app that places a call (creating a new channel) and bridges upon answer |
| 13:49.28 | klaxa|work | what is "app" in this context? |
| 13:49.37 | newtonr | klaxa|work, https://wiki.asterisk.org/wiki/display/AST/Asterisk+Architecture%2C+The+Big+Picture |
| 13:49.40 | newtonr | klaxa|work, https://wiki.asterisk.org/wiki/display/AST/Types+of+Asterisk+Modules |
| 13:49.45 | file | AppDial is just a name that Dial() fills in on any outgoing calls it creates |
| 13:49.47 | newtonr | Those two pages will help a little bit |
| 13:50.30 | [TK]D-Fender | the AppDial() in there is because that channel was created by a Dial(). This was not a channel that simply arrived at your server and is executing dialplan of its own |
| 13:51.05 | [TK]D-Fender | the calling end is the one that is executing dialplan and may continue on afterwards, etc |
| 13:52.06 | klaxa|work | hmm okay, that made it a bit cleary, still kinda fuzzy to me, i'll first read those pages and if i'm still confused try to come up with good questions, thanks! |
| 13:52.12 | klaxa|work | *clearer |
| 13:54.11 | *** join/#asterisk irc08153 (d9f7dd51@gateway/web/freenode/ip.217.247.221.81) |
| 13:55.25 | irc08153 | sip show peers, shows IP-PROVIDER-153333757558 195.185.37.60 N 5064 OK (29 ms), but tcpdump shows "15:42:30.018532 IP askoziapbx.local.5060 > 195.185.37.60.5060: SIP: REGISTER sip:sip.easybell.de SIP/2.0" Why? |
| 13:55.46 | irc08153 | not honoring the port |
| 13:57.32 | Samot | irc08153: Is the PBX behind NAT? |
| 13:57.58 | irc08153 | yes, but the tcpdump was taken from the pbx itself |
| 13:58.21 | Samot | P-PROVIDER-153333757558 195.185.37.60 N 5064 OK (29 ms) <-- 5064 is the port the other side is listening on |
| 13:58.25 | Samot | Not you. |
| 13:58.26 | Samot | Them. |
| 13:58.52 | Samot | They register to you on 5060 because that's what you are listening on |
| 13:59.02 | Samot | Their device/system is listening on 5064. |
| 13:59.07 | Samot | Not uncommon. |
| 13:59.14 | irc08153 | yes, but i should address the register to port 5064 |
| 13:59.22 | Samot | No. |
| 13:59.24 | Samot | It's THEIR port. |
| 13:59.35 | Samot | They are telling Asterisk, send stuff to this IP and Port |
| 13:59.50 | Samot | Just like you told the device send stuff to Asterisk on this IP and port. |
| 14:00.05 | [TK]D-Fender | irc08153, those 2 things have nothing to do with each other |
| 14:00.19 | [TK]D-Fender | irc08153, One is a PEER. The other is a REGISTER |
| 14:00.30 | [TK]D-Fender | irc08153, they are 5000% separate from one another |
| 14:01.22 | [TK]D-Fender | <irc08153> yes, but i should address the register to port 5064 <- go fix your register string |
| 14:01.42 | irc08153 | "<Samot> Their device/system is listening on 5064." |
| 14:01.45 | irc08153 | for what kind of packages? |
| 14:02.00 | [TK]D-Fender | <[TK]D-Fender> <irc08153> yes, but i should address the register to port 5064 <- go fix your register string |
| 14:02.12 | igcewieling | irc08153: are you on windows or linux? |
| 14:02.39 | irc08153 | linux |
| 14:02.47 | [TK]D-Fender | igcewieling, simple config mistake. Let him fix it |
| 14:03.02 | irc08153 | i want to understand it, and dont have to fix anything |
| 14:03.19 | [TK]D-Fender | it's not sending to the port you want because you didn't tell it to |
| 14:03.24 | irc08153 | everything is working and fine |
| 14:03.51 | *** join/#asterisk bford (d8cff501@gateway/web/freenode/ip.216.207.245.1) |
| 14:04.14 | [TK]D-Fender | It's sending to 5060 and you asked why it wasn't sending to 5064... |
| 14:04.21 | igcewieling | irc08153: now you know the port shown in "sip show peers" isn't very useful. |
| 14:04.22 | [TK]D-Fender | That'd be because you didn't tell it to |
| 14:04.40 | igcewieling | move on with your life. |
| 14:04.48 | irc08153 | funny guys |
| 14:07.02 | irc08153 | so to register my numbers at provider, i see "askoziapbx.local.5060 > 195.185.37.60.5060" i would expect port 5064 |
| 14:07.44 | irc08153 | as defined in sip.conf register => "....":"..."@sip.easybell.de:5064 |
| 14:07.57 | irc08153 | isnt that the "destination" to register? |
| 14:08.01 | igcewieling | irc08153: the provider is listening on port 5060, why would you want to connect to 5064? |
| 14:09.24 | irc08153 | i dont get it, why register =>... is defnied with port 5064 when the register package itself is DST Port 5060 |
| 14:09.58 | igcewieling | irc08153: I don't know why asterisk isn't using your bad setting. |
| 14:10.29 | igcewieling | maybe you forgot to issue a reload after making the change? |
| 14:11.13 | irc08153 | so we agree, if yo u set in sip.conf register => "....":"..."@sip.easybell.de:5064 - the register package itself should also go to port 5064 right? |
| 14:11.51 | *** join/#asterisk kharwell (kharwell@nat/digium/x-mtczvikxbmsjbkrj) |
| 14:11.51 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:11.59 | igcewieling | irc08153: assuming the stuff you are hiding it correct, then yes it should send to port 5064 |
| 14:12.22 | igcewieling | Did you issue a "sip reload" or "core reload" after making changes to sip.conf? |
| 14:12.23 | irc08153 | ok, thats all i wanted to know as i assumed a bug in the pbx gui |
| 14:12.29 | igcewieling | GUI? |
| 14:12.40 | irc08153 | its askozia as you should have seen by the hostnames |
| 14:12.58 | irc08153 | i wanted to verify a bug |
| 14:13.01 | igcewieling | damnit, bamboozled again! Very little of what I said applies when using Asterisk GUIs. |
| 14:13.11 | irc08153 | doesnt matter |
| 14:13.21 | Samot | http://askozia.com/support/ |
| 14:13.31 | irc08153 | you guys make me smile |
| 14:13.35 | Samot | Sorry man |
| 14:13.40 | Samot | We don't know how they have done the PBX |
| 14:13.46 | irc08153 | i dont want any support, just want to verify my thoughts |
| 14:13.47 | Samot | Which means what we say may not apply. |
| 14:13.57 | irc08153 | it feels like battling in here all the time |
| 14:14.01 | *** join/#asterisk ruied (~ruied@81.84.234.209) |
| 14:14.07 | Samot | Does sip.easybell.de want you to register on 5064 |
| 14:14.08 | Samot | ? |
| 14:14.17 | Samot | That's the port they use? |
| 14:14.31 | irc08153 | why are you al lthe time questioning setups? i asked simple questions about registration process and ports |
| 14:14.39 | Samot | Yes. |
| 14:14.46 | Samot | If you put the ;port it will go to that port |
| 14:14.50 | Samot | So what is the issue? |
| 14:15.28 | irc08153 | that its present in sip.conf but not honored, so my assumptions are correct, if it's there and "loaded" it should register at dst port 5064 not port 5060 as it does right now |
| 14:15.39 | irc08153 | so now i can adress that/forward that to askozia support |
| 14:15.42 | Samot | asterisk -rvvvvv |
| 14:15.47 | Samot | sip set debug on |
| 14:15.52 | Samot | Show register attempt to them |
| 14:15.58 | Samot | !pb |
| 14:15.59 | Samot | ~pb |
| 14:15.59 | infobot | hmm... 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:16.55 | igcewieling | Samot: his issue appears to be that he set the register line to register to 5064, but it still registers to 5060. |
| 14:16.56 | irc08153 | its already clear from tcpdump |
| 14:17.09 | Samot | I want to see a full debug |
| 14:17.09 | irc08153 | no reason to add additonal overhead |
| 14:17.20 | Samot | I want to see the full REGISTER packet from Asterisk. |
| 14:17.56 | Samot | Well you can show it now. |
| 14:18.03 | Samot | Or you can contact their support and show it to them. |
| 14:18.06 | Samot | Either way. |
| 14:18.08 | Samot | you're showing it. |
| 14:19.10 | irc08153 | seriously? tcpdump is a few layers lower |
| 14:19.16 | irc08153 | what do you expect, its in sip debug also port 5060 |
| 14:19.27 | irc08153 | https://pastebin.ca/3836683 |
| 14:20.03 | igcewieling | irc08153: sip set debug on shows us how ASTERISK sees the packets, not the OS. |
| 14:20.22 | Samot | Show your peer settings and mask the secret. With the register string. |
| 14:20.24 | Samot | Yes. |
| 14:20.46 | [TK]D-Fender | igcewieling, we're backwards |
| 14:20.47 | [TK]D-Fender | Reliably Transmitting (NAT) to 195.185.37.60:5060: |
| 14:20.54 | Samot | I was going to cover that |
| 14:20.57 | [TK]D-Fender | We're talking DST PORT here |
| 14:20.59 | Samot | I wanted to see the peer settings |
| 14:21.06 | [TK]D-Fender | peer = IRRELEVENT |
| 14:21.09 | igcewieling | I could ask what version of Asterisk he is using and mess everything up. |
| 14:21.50 | irc08153 | thank you guys for verifying my thoughts, time for askozia to jump in |
| 14:21.56 | [TK]D-Fender | nope |
| 14:22.08 | [TK]D-Fender | this is entirely on your register string |
| 14:22.21 | Samot | 10:20:24 AMÂ <Samot>Â Show your peer settings and mask the secret. With the register string. |
| 14:22.49 | [TK]D-Fender | <[TK]D-Fender> peer = IRRELEVENT |
| 14:22.50 | irc08153 | that comes from the askozia gui, thats nothing touchable by users |
| 14:23.07 | [TK]D-Fender | and is the line it made RIGHT? |
| 14:23.16 | [TK]D-Fender | If not, then it's their problem |
| 14:23.27 | igcewieling | So, what version of Asterisk does this GUI use? |
| 14:23.27 | [TK]D-Fender | (assuming you did everything you were supposed to from your end) |
| 14:23.56 | *** join/#asterisk pawiecki (~pawiecki@router.dir.pl) |
| 14:26.11 | pawiecki | Hi, I know it's maybe not 100% on topic, but I need some advice. I have this old KWS6000 DECT system, with ~60 Base stations and ~90 Handsets. It wasn't deployed in a proper way, and my job now is to fix some minor errors. Right now I'm looking to re-do the sync chain, to have it's source in the middle of the system and better sync tree. It's a 3 floor building, and I'm not sure how to plan sync chain - make it floor-by-floor or jump-around-the-best-s |
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| 14:26.13 | pawiecki | ignal-strength type? If you have some experience with this sort of thing, any advice would be appreciated. |
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| 14:33.37 | Samot | pawiecki: I don't think you're going to get a bite no matter how many times you cast the line. |
| 14:34.05 | Samot | That's a very specific type of setup with a very specific set of hardware. |
| 14:34.51 | Samot | Even then, you're hoping someone has not used used KWS6000's but have dealt with sync chains. |
| 14:34.55 | Samot | -not |
| 14:41.50 | pawiecki | Samot: yeah i know, sorry for repeating. It's pretty well documented on how to deploy it from the grounbd up, but not really on how to make changes to a working system, that's not setup right. |
| 14:42.22 | Samot | Oh the repeating wasn't an issue. |
| 14:42.50 | Samot | You asked _hours_ ago, so really it was fine. |
| 14:43.02 | Samot | I was just saying that you'll probably not get a bite. |
| 14:43.25 | Samot | It's a very narrow field you're looking in. |
| 14:45.31 | Samot | If I knew anything about those devices, I'd help but I've never used them. |
| 14:45.54 | pawiecki | They seem to be rare here too. |
| 14:47.42 | [TK]D-Fender | This is *. Asking about some DECT toaster .... is like trying to order a Bic Mac from Wendy's. |
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| 14:52.08 | pawiecki | [TK]D-Fender: thanks sir, you just made me hungry. Anyway, no more offtopic from me :) |
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| 15:20.17 | *** join/#asterisk Alonzo (6cb077da@gateway/web/freenode/ip.108.176.119.218) |
| 15:21.56 | Alonzo | hello everyone, i was wondering if anyone knew of a simple way to make a dialplan pattern that would accept calls from +1NPANXXXXX, 1NPANXXXXXX, and NPANXXXXXX without having to make separate dialplans for each type? |
| 15:24.56 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1::1:3) |
| 15:33.02 | igcewieling | Alonzo: you could do it in 2 patterns, but not 1. |
| 15:34.41 | Alonzo | would one pattern handle +1NPANXXXXXX and 1NPANXXXXXX? |
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| 15:35.44 | igcewieling | no |
| 15:35.52 | igcewieling | sorry, yes |
| 15:36.09 | igcewieling | _[+1]NXXNXXXXXX |
| 15:36.46 | Alonzo | i see, i will give that a shot. Thank you for the assistance |
| 15:36.52 | igcewieling | On my dialplans I simply have 3 lines |
| 15:37.12 | igcewieling | silly to make things more complicated |
| 15:38.45 | Alonzo | the problem is that we have over 10K DIDs and we have separate entries to handle each type of incoming DID (+1,1, neither) and it really slows down reloads. I was trying to see if we could remove a large portion of entries by optimizing the dialplan pattern |
| 15:39.14 | Alonzo | from what you suggested that will help already |
| 15:39.43 | igcewieling | exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) |
| 15:39.59 | [TK]D-Fender | <igcewieling> sorry, yes < -no |
| 15:40.12 | igcewieling | exten => _+NXXNXXXXXX,1,Goto(1${EXTEN:0},1) etc. |
| 15:40.44 | igcewieling | [TK]D-Fender: : _[+1]NXXNXXXXXX won't work? |
| 15:40.48 | [TK]D-Fender | no |
| 15:40.58 | [TK]D-Fender | [] <- single digit list of possible values |
| 15:41.10 | igcewieling | correct. match leading + OR leading 1 |
| 15:41.16 | [TK]D-Fender | <Alonzo> would one pattern handle +1NPANXXXXXX and 1NPANXXXXXX? |
| 15:41.22 | [TK]D-Fender | And his request had a one in BOTH |
| 15:41.23 | igcewieling | Im not trying to match +1 |
| 15:41.36 | [TK]D-Fender | His + was optional |
| 15:41.41 | [TK]D-Fender | Not the same length |
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| 17:20.53 | Alonzo | igcewieling: It appears that that pattern doesn't work. I am getting "rejected because extension not found in context " messages with +1NPANXXXXXX invites. Do you think there is a tweak that can be made to it? |
| 17:24.34 | [TK]D-Fender | no |
| 17:24.39 | [TK]D-Fender | You need separate patterns |
| 17:24.59 | [TK]D-Fender | at least 1 line per pattern with a Goto() as he gave you a sample for |
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| 18:12.17 | hdon | hi all :) does channel variable inheritance occur when a channel is instantiated or are they synchronized between channels until they are destroyed? |
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| 18:17.08 | file | instantiated. |
| 18:18.14 | hdon | thanks file |
| 18:18.20 | hdon | cool nick |
| 18:19.58 | hdon | when does asterisk write to the CDR? |
| 18:23.02 | [TK]D-Fender | when the call is done |
| 18:23.29 | [TK]D-Fender | there is a rule that can be set to determine if that happens as "h" gets called (or similar), or until the absolute end |
| 18:23.33 | [TK]D-Fender | (IIRC) |
| 18:27.38 | hdon | hmm |
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| 19:43.55 | drmessano | 14:18:20 <hdon> cool nick |
| 19:43.57 | drmessano | ^ No |
| 19:44.05 | drmessano | The coolest nick is 'nick' |
| 19:44.39 | drmessano | Plus, there's a rule about feeding file's ego and stuff |
| 19:44.55 | drmessano | Apparently he can be fed muffins, but not compliments |
| 19:45.01 | drmessano | Super old rule |
| 19:46.05 | file | I don't have time for ego feeding |
| 19:48.03 | drmessano | file: Understood, BTW cool nick |
| 19:48.32 | file | <PROTECTED> |
| 19:48.47 | Samot | My ancestors owned a nick. |
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| 20:58.19 | Kobaz | mmmm |
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| 21:59.03 | rp_ | Has anyone had an issue with including context's in another context? I'm running Asterisk 13.13.1 and do have some troubles with including context's in another context in my dialplan. I do only see the problem if it's a normal "all digit" extension. If the included extension starts with a * it does work as planned. Any ideas regarding my issue? |
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| 22:14.15 | [TK]D-Fender | Inclues work |
| 22:14.26 | [TK]D-Fender | I've never a single but in this thing before |
| 22:14.57 | [TK]D-Fender | You've done something wrong. Show us your dialplan and a call and we'll show you where |
| 22:15.00 | [TK]D-Fender | ~pb |
| 22:15.01 | infobot | extra, extra, read all about it, 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 22:15.02 | [TK]D-Fender | ^^^ |
| 22:16.34 | rp_ | [TK]D-Fender: Comming up |
| 22:17.17 | [TK]D-Fender | bug* |
| 22:20.42 | rp_ | [TK]D-Fender: You can find what I think is relevant here: https://pastebin.com/F2FD7zSC |
| 22:21.09 | rp_ | I cna dial *31* or *99* but I can't dial 8000 |
| 22:24.30 | rp_ | [TK]D-Fender: You find a test call here: https://pastebin.com/APZaZ2Em |
| 22:29.55 | salviadud | rp_, your last line should be Hangup() but it says Hanup() |
| 22:30.11 | salviadud | Not that it will fix your main problem though |
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| 22:32.15 | Samot | exten => _X.,n,GotoIf($["${ODBC_ANONYM-LOOKUP(${CALLERID(num)},C030995)}" = "1"]?features,*31*${EXTEN},1) <-- What is there an * before the ${EXTEN}? |
| 22:33.04 | Samot | Oh I see how you're doing it, it was at the bottom. |
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| 22:37.32 | Samot | Where's the RealTime dialplan to look at? |
| 22:39.30 | [TK]D-Fender | [Jun 30 00:06:21] -- Executing [8000@C030995-phones:1] Set("SIP/C030995-01884651-00000047", "CENTRALID=C030995") in new stack |
| 22:39.35 | [TK]D-Fender | I see 8000 dialed... |
| 22:40.27 | Samot | [Jun 30 00:06:21] -- Executing [8000@outbound:1] Dial("SIP/C030995-01884651-00000047", "SIP/8000@production-sw1-s01") in new stack |
| 22:41.45 | [TK]D-Fender | Where do I see 8000 failing? |
| 22:43.01 | Samot | Well [outbound] is the only context not shown in the pb's. And there are calls to RT dialplan.. |
| 22:43.17 | Samot | 6:37:35 PM S<Samot> Where's the RealTime dialplan to look at? <-- So back to my question... |
| 22:44.25 | Samot | [Jun 30 00:06:21] -- Executing [8000@outbound:1] Dial("SIP/C030995-01884651-00000047", "SIP/8000@production-sw1-s01") in new stack <-- And how come the debug ends here? What is the dial attempts and results of that call? |
| 22:44.35 | Samot | s/What/Where/ |
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| 23:34.44 | hdon | drmessano, :3 |
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| 23:38.14 | lorsungcu | :3 |
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