00:01.22 | [TK]D-Fender | And saying that doesn't add anything to the conversation. No change in methods, no testing, no added description about what's on each end of the call. Nothing. |
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00:01.53 | [TK]D-Fender | I recommend correcting any (and or ALL) of these points. |
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01:02.57 | Katty | Bloop |
01:04.38 | [TK]D-Fender | Katty, Mew. |
01:08.12 | Katty | Herro fenderbender |
01:10.31 | [TK]D-Fender | MsPrettyKittyKatty |
01:10.34 | [TK]D-Fender | Howgoesit? |
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01:36.26 | Katty | It goes. You? |
01:36.44 | Katty | Any major life changes in the last 1.5yrs? |
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01:49.03 | [TK]D-Fender | Nope. |
01:49.50 | [TK]D-Fender | SSDD |
01:57.04 | Katty | Good! |
01:57.59 | Katty | Not much change here either. Moved into a town house in April. Probably start looking for a house to buy in another year or two |
01:58.32 | [TK]D-Fender | Slow & steady keeps up with the glaciers |
02:00.37 | Katty | Yep |
02:02.11 | [TK]D-Fender | Upgrading homes is a thing... |
02:04.46 | Katty | Is it? |
02:06.23 | [TK]D-Fender | yup. Change of lifestyle with more property upkeep, independent taxes, full control, full responsibility |
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02:14.44 | Katty | Sounds like adulting is involved |
02:17.05 | Katty | Trying to keep the adulting to a minimum |
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07:45.45 | DanQuinney | adulting sucks |
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09:18.23 | wyoung | hey gang |
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12:24.57 | kim_bruning | hello, in my sip tcpdump connecting to xs4all, in a REGISTER request, I get From: <sip:0858769839@sip.xs4all.nl>;tag=as7bad8b24 ... To: <sip:0858769839@sip.xs4all.nl> my version is User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 |
12:25.03 | kim_bruning | is this actually what's supposed to be happening |
12:25.31 | kim_bruning | I have a 2nd provider as well, and that logs in differently, so I'm a tad confuzzeled how that's possible |
12:28.42 | wyoung | <PROTECTED> |
12:31.41 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
12:31.57 | [TK]D-Fender | kim_bruning, Donèt see why not. What is the actual problem? |
12:32.17 | wyoung | [TK]D-Fender: Lets pal it |
12:32.21 | wyoung | [TK]D-Fender: TF2? |
12:32.30 | wyoung | FTP |
12:38.03 | kim_bruning | I can't register with my upstream provider |
12:38.28 | kim_bruning | well, I could until yesterday. I may have unwittingly updated asterisk |
12:38.30 | [TK]D-Fender | Show us the full attempt |
12:39.05 | [TK]D-Fender | "my version is User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3" <- this is not what I'd call "updated" |
12:39.18 | [TK]D-Fender | 1.8 = DEAD. so is 10 & 12 |
12:39.18 | kim_bruning | debian |
12:39.39 | [TK]D-Fender | they have 11 in their repos |
12:39.44 | [TK]D-Fender | That is not an excuse |
12:40.00 | [TK]D-Fender | Also not a reason for a failure though... |
12:40.18 | [TK]D-Fender | just kills any support and means you're massively behind on fixes |
12:40.30 | kim_bruning | http://pastebin.com/TZFcXZXi |
12:40.36 | kim_bruning | suggest update to 11? |
12:40.45 | [TK]D-Fender | Full attempt from * CLI w/ SIP debug |
12:40.47 | [TK]D-Fender | not external |
12:40.48 | Nivex | I miss debian backport |
12:41.17 | kim_bruning | fair enough. let's see |
12:43.40 | kim_bruning | http://pastebin.com/TNbZFgez |
12:43.44 | kim_bruning | this what you're looking for? |
12:44.13 | kim_bruning | or if not, what should I be doing instead? |
12:46.00 | kim_bruning | the cli version looks very similar to the tcpdump |
12:46.30 | [TK]D-Fender | And there is NO response from them at all? |
12:46.41 | kim_bruning | indeed not, they have a 401 in their logs |
12:46.55 | kim_bruning | and that's all they tell me |
12:47.01 | kim_bruning | and no response |
12:47.05 | [TK]D-Fender | then something is blocking that |
12:47.23 | [TK]D-Fender | check your firewalls on the server / routers & confirm your WAN IP on the contact |
12:47.31 | [TK]D-Fender | As well as any required forwarding |
12:47.33 | kim_bruning | you'd think so, though a different provider *is* responding correctly (and is -in fact- connecting) |
12:47.36 | [TK]D-Fender | Contact: <sip:s@82.94.48.148:5060> |
12:47.37 | [TK]D-Fender | ^ |
12:47.54 | kim_bruning | that port is open and responding |
12:47.57 | [TK]D-Fender | A different provider might ignore what you send for the return. |
12:48.07 | kim_bruning | could be! |
12:48.44 | [TK]D-Fender | Does look like the IP your IRC connection is from.... |
12:49.23 | [TK]D-Fender | There is some reason why the packet isn't making it back if they say they got the REGISTER and sent the 401. |
12:49.32 | [TK]D-Fender | ask for a confirmation of where they were sending |
12:49.36 | [TK]D-Fender | get their dump |
12:49.43 | [TK]D-Fender | and then start tracing backwards from there |
12:50.06 | kim_bruning | I already asked... and they didn't provide further info ^^;;; |
12:50.19 | kim_bruning | so trying other avenues first |
12:51.14 | kim_bruning | and traffic to port 5060 is coming through (tcpdump on this end, and remote box nc <ip> 5060 ) |
12:51.49 | kim_bruning | if I update to .11 , is it likely my config will bork? |
12:54.06 | kim_bruning | incidentally, if I test the connection with zoiper, I can immediately log in. |
12:54.13 | kim_bruning | (which I should have mentioned earlier) |
12:58.18 | *** join/#asterisk zaf (~zaf@76.72.92.37) |
12:58.21 | [TK]D-Fender | Somehting else is wrong |
12:58.22 | [TK]D-Fender | and itès not the * version |
12:59.02 | whizzi | kim_bruning: Do you get any reply from Xs4all from your register attempt? |
12:59.49 | kim_bruning | whizzi: I do not, not in asterisk |
13:00.26 | whizzi | then thereâs a network related issue |
13:01.08 | [TK]D-Fender | [TK]D-Fender> then something is blocking that |
13:01.08 | [TK]D-Fender | <[TK]D-Fender> check your firewalls on the server / routers & confirm your WAN IP on the contact |
13:01.14 | whizzi | either NAT problem or a too strict firewall that doesnât allow RELATED connections to return |
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13:01.31 | whizzi | [TK]D-Fender as always, you are right ;-) |
13:02.08 | whizzi | Xs4all will see a 401 being sent to your box, but that one never comes through |
13:02.28 | whizzi | this 401 has the nonce code in it, allowing Asterisk to register using this code |
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13:02.43 | whizzi | if thatâs okay, combined with the password, you are registered |
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13:02.50 | kim_bruning | right |
13:03.19 | kim_bruning | <scratches head> |
13:03.22 | Phrohdoh | Hello everyone, I just discovered asterisk late yesterday and I only have a single question so far. Is it possible for one user to directly leave another user a voicemail? |
13:03.49 | kim_bruning | my WAN IP is ok |
13:03.53 | zaf | Phrohdoh, yes |
13:04.07 | Phrohdoh | zaf: Excellent, thank you for the quick response. :-) |
13:04.33 | Phrohdoh | Does anyone here run asterisk on a raspi? I saw a project just for that yesterday and am curious to know if that is viable? |
13:04.59 | zaf | don't expect to run a call center on it, but for small installs, yes |
13:06.10 | [TK]D-Fender | Phrohdoh, It's your system, it'll do whatever you configure it to do. |
13:06.10 | Phrohdoh | Great to hear. I would only throw it on my pis so that I may learn the system a bit. Of course a full install on a much more suited machine will follow. Thanks! |
13:06.48 | Phrohdoh | [TK]D-Fender: While I don't doubt that I have _never_ done anything with a system like Asterisk. |
13:06.58 | [TK]D-Fender | pi = puny. If you only inttend ona tiny # of connection it could do the job if you aren't transcoding, etc |
13:07.10 | [TK]D-Fender | ~book |
13:07.10 | 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 |
13:07.12 | [TK]D-Fender | ^^^ |
13:07.19 | [TK]D-Fender | Time to read up & learn about it |
13:07.24 | Phrohdoh | Great I will do that! |
13:08.22 | Phrohdoh | Hm the online version of the book focuses on 1.8 but given the title for this channel I assume that is quite dated. Would that be correct? |
13:08.57 | zaf | 4th edition is more up to date |
13:10.51 | Phrohdoh | Thanks again - http://asterisk-service.com/downloads/Asterisk-%20The%20Definitive%20Guide,%204th%20Edition.pdf |
13:21.00 | kim_bruning | well, turning nat on, means I have xs4all listed in my peers |
13:21.02 | kim_bruning | that's a start |
13:22.12 | kim_bruning | asterisk sends a message, then gets a reply from xs4all |
13:22.26 | kim_bruning | SIP/2.0 200 OK |
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13:22.41 | kim_bruning | but can't call out for some reason <scratches head> |
13:23.06 | [TK]D-Fender | Considered showing us? |
13:23.28 | kim_bruning | can call in |
13:23.30 | kim_bruning | hang on |
13:23.44 | whizzi | *keeps hanging* |
13:23.55 | kim_bruning | ok, what would you like to see |
13:24.01 | [TK]D-Fender | grabs a big stick |
13:24.14 | kim_bruning | call in works, I figure I probably left something silly in the sip.conf now |
13:24.24 | [TK]D-Fender | #piñatatime |
13:25.22 | kim_bruning | http://pastebin.com/jLUxN0nB |
13:25.31 | kim_bruning | I'm showing, I'm showing! |
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13:28.16 | whizzi_ | see what you did [TK]D-Fender |
13:28.51 | wyoung | hi |
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13:30.23 | whizzi | kim_bruning: Do you see an attempt calling out on the xs4all trunk? |
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13:31.08 | kim_bruning | <PROTECTED> |
13:32.10 | whizzi | check your from-header, rpid and callerid |
13:32.42 | [TK]D-Fender | kim_bruning, Where's the CALL? |
13:34.14 | Phrohdoh | So I think I got some misinformation yesterday. Asterisk itself is not a distribution of GNU/Linux, it is instead a suite of tools that were built to run on a linux system, correct? |
13:36.01 | kim_bruning | Phrohdoh: correct |
13:36.14 | kim_bruning | geh, I'm getting traffic in between |
13:38.00 | [TK]D-Fender | Yes, but there are distros made exclusively for the purpose of deploying an * based system |
13:38.05 | [TK]D-Fender | Mostly GUI driven though |
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13:42.34 | kim_bruning | I have sip debug on .... |
13:42.40 | kim_bruning | so ... trying to figure where this started |
13:44.19 | [TK]D-Fender | reboots to take updates.... |
13:45.45 | kim_bruning | http://pastebin.com/Gr9Wf9Bf |
13:45.48 | kim_bruning | this should cover it |
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13:56.02 | [TK]D-Fender | kim_bruning, you aren't showing the INVITE tiself |
13:56.26 | kim_bruning | I might have gotten it now |
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14:00.59 | kim_bruning | http://pastebin.com/up0Htrk6 |
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14:05.29 | [TK]D-Fender | ^MESC[Kvosserver*CLI> ^MESC[0KReliably Transmitting (NAT) to 194.109.16.16:5060: |
14:05.36 | [TK]D-Fender | Your provider should not be behind NAT |
14:06.17 | kim_bruning | preferably ^^;; |
14:06.20 | [TK]D-Fender | but that isn't the immediate problem |
14:06.27 | kim_bruning | *nod* |
14:06.34 | Phrohdoh | In the context of this project what is a 'trunk'? |
14:06.38 | [TK]D-Fender | Thre is another auth issue |
14:06.57 | [TK]D-Fender | Phrohdoh, A vague term often misused |
14:07.37 | [TK]D-Fender | Phrohdoh, But typically meant to be a tech that allows you to connect to the PSTN for in and/or outbound calls to the PSTN |
14:08.11 | kim_bruning | D-Fender: I got to the point where I can see it's 403ing me when I try to connect |
14:08.21 | kim_bruning | but I'm not entirely sure why |
14:08.29 | [TK]D-Fender | Where "SIP trunk" is used to describe call/registration auth setup to use an account with a SIP provider |
14:08.40 | [TK]D-Fender | the term also applied to wired connections |
14:08.45 | Phrohdoh | Thanks, that makes a little more sense. That is quite vague though. |
14:08.47 | [TK]D-Fender | like analog and digital land-lines |
14:09.30 | [TK]D-Fender | It is sometimes used in terms of talking aboutt the receiving end specifically, other times referring toa config setup on your side |
14:09.38 | [TK]D-Fender | it's all aboutt context |
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14:14.18 | Phrohdoh | Which distro would you all suggest to host an asterisk installation? I could spin up a vm on my current machine, but I wouldn't be able to dedicate any 'real' amount of resources to that, nor can I afford another machine currently. |
14:16.05 | Phrohdoh | I should have prefaced that with what I'd like to build. |
14:16.20 | Phrohdoh | (note that I am reading the book currently, too) |
14:17.00 | kim_bruning | here's my config. am I doing something obviously wrong? http://pastebin.com/hah3ntTm |
14:17.53 | kim_bruning | continues looking |
14:18.07 | Phrohdoh | I asked about voicemail earlier and would like to build a system so that a user may send another user a voicemail directly. But ideally the wav would be prerecorded so the user wouldn't have to actually speak, just select a wav and a recipient. Of course the user interactions could be built into a web interface, or a command line tool, etc (which i |
14:18.08 | Phrohdoh | sn't directly related to asterisk, I assume). So my understanding is that I require very little to do this. |
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14:22.45 | kim_bruning | pokes the isp |
14:23.38 | kim_bruning | D-fender: hmm, anything I can read up on or track down ? ^^;; |
14:23.50 | [TK]D-Fender | Phrohdoh, Everything depends on your needs. the GUI distros offer basic cookie-cutter approaches to a variety of aspects of traditional closed PBX's |
14:24.05 | kim_bruning | Phrohdoh: I'm told debian not so great ;-) |
14:24.30 | [TK]D-Fender | this includes how you dial between phone devices (treated as "people"), processing inbound vs outbound calls, queues, IVRs , etc |
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14:24.39 | kim_bruning | (because it installs a really old asterisk) |
14:24.43 | [TK]D-Fender | kim_bruning, your decript PACKAGES are not great |
14:24.54 | [TK]D-Fender | kim_bruning, And they at least HAVE 11 available... but you're not using that |
14:25.06 | kim_bruning | <PROTECTED> |
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14:25.42 | kim_bruning | I'll be testing that on a test box later |
14:26.11 | kim_bruning | though currently I'm working through auth stuff |
14:28.45 | Phrohdoh | [TK]D-Fender: Well, I understand that everything depends on my needs but that is why I have laid them out. I am probably not asking the best questions because I am completely new to dealing with a PBX/Asterisk. My day job is a health insurance platform and I'd like to integrate automated voicemail sending under certain conditions. Of course asteris |
14:28.45 | Phrohdoh | k wouldn't be aware of the consumer's conditions. I'll do some more reading over the week. |
14:30.32 | Phrohdoh | kim_bruning: Ha ok thanks. I'd like to not go with a cookie-cutter GUI-oriented solution because I plan to really learn the tools and eventually customize them as my vision changes, but perhaps that'll be the easiest way for me to get started. |
14:31.15 | Phrohdoh | I am heading out for now though. Thank you for all the help so far! |
14:31.30 | [TK]D-Fender | Phrohdoh, " integrate automated voicemail sending under certain conditions" <- you'll want to express this call flow in proper detail to see if the GUI offers the means to your goals including this and to determin which paths might serve you best |
14:38.25 | kim_bruning | hmm, asterisk is sending local ip over net instead of wan ip <scratches head> |
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14:57.44 | seik0 | Hi! Is it possible to translate DIALSTATUS and HANGUPCAUSE when we do Dial(Local/localchan-ext), and in localchan-ext we do "real" Dial(SIP/123) ? |
14:58.20 | seik0 | Because now when Dial(SIP/123) returns DIALSTATUS=NOANSWER, what to do next? |
14:59.10 | seik0 | in main dialplan DIALSTATUS became CHANUNAVAIL, and HANGUPCAUSE=0 |
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15:03.59 | seik0 | but maybe in newer asterisk releases this behaviour has already changed |
15:04.14 | seik0 | I use old 1.4 version |
15:12.08 | kim_bruning | so setting nat=no now works too? |
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15:13.03 | kim_bruning | it's gotta be something really simple, and I'm just not seeing it ^^;;;; |
15:18.37 | [TK]D-Fender | seik0, You'd have to show us an actual call to see what's happening. |
15:18.53 | [TK]D-Fender | seik0, And then .... you know you should not be on anything other tthan 11 or 13 by now |
15:19.14 | [TK]D-Fender | kim_bruning> so setting nat=no now works too? <- depends WHERE |
15:19.37 | seik0 | we talked about old versions at least 5 times here already =) |
15:19.48 | seik0 | but this is not as simple |
15:21.05 | [TK]D-Fender | I guess it's a question of how long you can stay on an ancient insecure and unsupported platform. |
15:21.10 | seik0 | many things stopped working when we tried to jump from 1.8 to 10, few days were not enough to fix all, so it was delayed till better days ) |
15:21.24 | [TK]D-Fender | And every year you put it off the situation gets worse and stupider to be in. |
15:21.46 | [TK]D-Fender | 1.8 = DOA. 10 = DOA |
15:22.06 | [TK]D-Fender | Better days never happen |
15:22.11 | [TK]D-Fender | You're still where you started |
15:23.08 | seik0 | reality is cruel |
15:23.42 | [TK]D-Fender | We keep saying "later". That's not reality, that's a conscious choice not to do the work. |
15:24.00 | [TK]D-Fender | either way... if you want any input on whatt you've got... then show it |
15:24.26 | seik0 | no, it is choise of kind "there are more important things to do", and even better idea of "do not touch while it works" |
15:25.30 | Nivex | ahh, I love me some "there are more important things to do". Seems like the "new shiny" always ends up being more important. The question is: when your phone system comes crashing down, will you get fired? That's how important it is. |
15:26.23 | kim_bruning | you know what would be funny? If my sip.conf was actually perfectly in order, and the actual bug was in extensions.conf |
15:27.04 | [TK]D-Fender | As in not dialing the correct format of # in the first place? |
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15:30.42 | seik0 | Nivex, when system is crashing - it is one thing. When for new asterisk release you need new version of drivers (and it is very serious), new OS kernel (and new OS all the way) - it is another. |
15:31.46 | seik0 | ok, let's stop for now |
15:32.50 | [TK]D-Fender | <[TK]D-Fender> either way... if you want any input on whatt you've got... then show it |
15:34.42 | seik0 | ok, ok, i'm working |
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16:03.13 | seik0 | It's "pseudo", just simplified description, because making it to work in real dialplans may take few hours of work: http://pastebin.com/yuSYgL89 |
16:04.22 | seik0 | is it ok? |
16:07.23 | [TK]D-Fender | You can't be getting "noanswer" from that |
16:07.39 | [TK]D-Fender | and waiting an hour for broken pseudo-code was a waste of everyone's time |
16:12.48 | seik0 | what do you mean "I can't be getting "noanswer"" ? Because channels was answered? |
16:14.45 | seik0 | mainly the question is whether is is possible to translate HANGUPCAUSE out from Local channel, and this pseudo-dialplan is just an explanaition of what I mean |
16:15.41 | seik0 | so for first to dive into making real plans I asked if it is expected to work or not |
16:16.16 | seik0 | if it is not expected, then you can always say that it may have been already fixed or smth |
16:17.41 | seik0 | btw, I already found some patches that sets hangupcause for local channel, though I expect some caveats |
16:19.57 | [TK]D-Fender | You use Hangup() to set the pass-back for this, not messing with the variable directly |
16:22.56 | seik0 | it works for for IAX2 call, but doesn't for call to Local channel |
16:23.29 | [TK]D-Fender | Come back with something real to compare |
16:23.40 | seik0 | ok, I'll be back then |
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17:08.41 | igcewieling | seik0: look at https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance and "core show function MASTER_CHANNEL" |
17:08.54 | igcewieling | that might help with chan_local stuff, I don't know. |
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23:52.32 | Katty | Hello my asterisk does not work at all, how to fix |
23:55.54 | WIMPy | hands Katty a light |
23:55.57 | WIMPy | Burn it |
23:56.13 | Katty | <PROTECTED> |
23:57.48 | WIMPy | Who knows. Using smoke signals might become fashionable again. |