| 00:19.22 | *** join/#asterisk matrix1233 (~matrix123@197.3.206.61) |
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| 02:16.25 | craigify | I'm really digging ARI |
| 02:20.51 | freakyfrank12 | sup |
| 02:21.24 | freakyfrank12 | What's the largest asterisk networks you've set up? |
| 02:25.48 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-rzsejmxfdoirqpgh) |
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| 02:59.42 | voipmonk | 18,000+ employees for one company |
| 02:59.59 | voipmonk | 20,000 subscribers for an older company |
| 03:00.08 | *** join/#asterisk Nate_15329 (~Nate15329@99-72-98-70.lightspeed.oshkwi.sbcglobal.net) |
| 03:00.08 | voipmonk | i haven't done anything beyond that yet. |
| 03:00.23 | voipmonk | freakyfrank12 |
| 03:05.49 | freakyfrank12 | dang |
| 03:11.51 | freakyfrank12 | I manage 1,500 devices for a medium sized school district, but we run CUCM. |
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| 06:11.08 | Vamp898 | When i receive an call from Asterisk, i always get the Call with @internalip, is there a way to change that? |
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| 06:13.31 | Vamp898 | In the configs, the internal IP is not specified, also the clients all connect to the Server via hostname (which resolves to the external ip) |
| 06:25.40 | Vamp898 | Or let me ask different, is there an config parameter which is responsible for setting the IP Adresse which comes after the @ |
| 06:58.27 | *** join/#asterisk st2 (~st@static.78-158-207-178.kgts.ru) |
| 07:02.06 | st2 | hello. i have a problem with outbound registration. it looks due to such log messages: |
| 07:02.07 | st2 | WARNING[27462]: sched.c:489 sched_settime: Bug likely: Negative time interval -49646 (interpreted as 4294917650 ms) requested! |
| 07:02.23 | st2 | any ideas how it could be fixed? |
| 07:04.52 | Vamp898 | st2: Just guessing but is the time on the server correct? |
| 07:05.29 | st2 | you mean ntp missmatching? |
| 07:06.13 | Vamp898 | st2: something like that, yes |
| 07:07.19 | *** join/#asterisk matrix1233 (~matrix123@197.3.229.93) |
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| 07:19.17 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 07:20.31 | st2 | but i have no idea with what should match time on server? date return correct value in my timezone |
| 07:21.08 | st2 | and 49646ms really strange difference |
| 07:24.22 | stefan27 | st2 which version |
| 07:24.24 | stefan27 | of asterisk |
| 07:24.53 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 07:26.11 | st2 | sorry, I should start from it ) 11.23.0 |
| 07:26.42 | st2 | in 11.20.0 it's work fine. i guess such warning was added in new version |
| 07:35.46 | *** join/#asterisk DanB (~DanB@178.138.98.61) |
| 07:41.06 | stefan27 | asterisk 13.9.0 had similar issue I think, then they quickly patched it to 13.9.1... That bug was due to using floating points to represent large integers |
| 07:41.26 | stefan27 | when the developers wake up you can ask what happend between 11.20 and 11.23 |
| 07:41.45 | stefan27 | It might be something entirely different of course; I have only used asterisk 13-13.5 |
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| 08:08.19 | st2 | stefan27 ok. thanks |
| 08:08.44 | st2 | i'll try to find out |
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| 11:22.42 | salz212 | Hi guys, is there something wrong with asterisk mailing list? I don't see any new email since Aug 24th. |
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| 11:25.50 | file | nope, nothing wrong |
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| 12:39.05 | st2 | hello, is there anybody of asterisk developers? i guess it is early morning now in your timezone) |
| 12:39.54 | file | some exist |
| 12:40.14 | file | your problem is different than stefan27 mentioned |
| 12:40.55 | file | nothing really comes to mind for what would cause it |
| 12:41.01 | st2 | morning) no the same problem. i've ask before it |
| 12:41.59 | st2 | asterisk will not produse outbound registration after "Negative time interval" log message |
| 12:42.55 | st2 | i didn't see such message on 11.20.0 asterisk. and haven't faced with this problem |
| 12:44.03 | *** join/#asterisk matrix1233 (~matrix123@197.3.229.93) |
| 12:44.50 | st2 | should i send bugreport about this issue? |
| 12:45.08 | file | yes - configuration, console output, everything you can |
| 12:45.34 | file | https://issues.asterisk.org/jira |
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| 12:48.28 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 13:13.48 | st2 | ok, thanks |
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| 13:18.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:02.14 | kchehab | hi ppl ,i have asterisk server running using tls :5161 , i buy a domain ,what to do to let my client register with domain name instead of my public ip |
| 14:02.56 | *** join/#asterisk garygraham (~Tadgh@216-26-192-156.dynamic.tbaytel.net) |
| 14:04.29 | TandyUK | setup dns? |
| 14:04.47 | TandyUK | also youll probably need a valud SSL cert |
| 14:05.23 | kchehab | TandyUK what do you mean by setup dns ,do you mean add domain to /etc/hosts ? |
| 14:06.16 | kchehab | in asterisk shell sip show settings From: Domain: Use domains as realms: No |
| 14:08.21 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
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| 14:24.39 | TandyUK | no as in setup dns with your domain provider / web host / whatever |
| 14:24.51 | TandyUK | so yourname.com resolves to your asterisk servers IP |
| 14:25.35 | TandyUK | you need a dns A record at minimum, and probably some SRV records as well |
| 14:26.02 | TandyUK | _sip._udp.yourdomain.com for eaxmple |
| 14:30.26 | kchehab | TandyUK yes my dns server ensure me this server |
| 14:30.43 | kchehab | but what should i configure in my asterisk |
| 14:30.56 | kchehab | fromdomain=yourname.com |
| 14:37.28 | *** join/#asterisk krzyzaq (~krzyzaq@user-94-254-226-38.play-internet.pl) |
| 14:37.48 | krzyzaq | hi all |
| 14:38.17 | krzyzaq | i have two quetions - can I post them? |
| 14:39.37 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-ihzupwvuknfeenqo) |
| 14:41.19 | WIMPy | We don't know about your abilities to ask, but |
| 14:41.24 | WIMPy | ~ask |
| 14:41.24 | 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. |
| 14:48.16 | krzyzaq | q1: where can I find a good tutorial how to set up asterisk with cisco cp-7961g phones and sip gateway? |
| 14:49.53 | krzyzaq | q2: is there a tutorial how to change the firmware of the cp-7961g? already tried some but the phone does'n want to read the files from tftp - something seeams to be missing. |
| 14:50.56 | [TK]D-Fender | <kchehab> fromdomain=yourname.com <- no need |
| 14:51.21 | WIMPy | For Cisco 79xx the S&M channel might be appropriate. |
| 14:51.49 | [TK]D-Fender | WIMPy, They're wired phones... that implies bondage too.... |
| 14:51.59 | *** join/#asterisk freakyfrank12 (~freakyfra@64.137.246.152) |
| 14:52.04 | WIMPy | True |
| 14:52.31 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 14:53.36 | *** part/#asterisk freakyfrank12 (~freakyfra@64.137.246.152) |
| 14:53.44 | [TK]D-Fender | krzyzaq, " and sip gateway" <- you'll have to be more specific |
| 14:53.50 | WIMPy | And to make things even worse the firmwae version can make quite some differences. |
| 14:54.59 | krzyzaq | so what is the recommended fw for those? |
| 14:56.11 | Chainsaw | krzyzaq: Well you're in the interesting situation that Asterisk supports SIP a lot better than SCCP. |
| 14:56.19 | Chainsaw | krzyzaq: And your chosen handset supports SCCP a lot better than SIP. |
| 14:57.24 | krzyzaq | Chainsaw: fortunately, just for now I don't have no need to use any sophisticated functions, just calling and receiving calls |
| 14:58.03 | Chainsaw | krzyzaq: You're still going to be using either SIP or SCCP to control those calls. |
| 14:58.07 | krzyzaq | by the sip gateway I understang a service that I bought from a local provider |
| 14:58.14 | WIMPy | That can be rather sophisticated already in that case. |
| 14:58.38 | krzyzaq | the provider is fcn.pl if it helps |
| 15:02.25 | [TK]D-Fender | It'll be 99% like any other provider |
| 15:02.27 | [TK]D-Fender | SIP is SIP |
| 15:02.56 | [TK]D-Fender | username, passwords, possibly"fromdomain" as required. Standard reg string, etc |
| 15:03.42 | krzyzaq | [TK]D-Fender: but is there any good explained tutorial? where, what, why? |
| 15:04.10 | [TK]D-Fender | The basics are just that... |
| 15:04.16 | [TK]D-Fender | the sample config explains each parameter. |
| 15:04.25 | [TK]D-Fender | that's the first thing. then to look at some real-world samples |
| 15:04.34 | [TK]D-Fender | so |
| 15:04.36 | [TK]D-Fender | ~book |
| 15:04.36 | 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 |
| 15:04.37 | [TK]D-Fender | ^ |
| 15:04.44 | [TK]D-Fender | as well as samples from OTHER providers |
| 15:04.48 | [TK]D-Fender | as a point of reference |
| 15:09.14 | *** join/#asterisk SunGod (~SunGod@92.55.116.125) |
| 15:13.32 | krzyzaq | ill check the book |
| 15:13.40 | krzyzaq | and about the fw? |
| 15:19.20 | [TK]D-Fender | Plenty of people offering blog posts etc for setting that phone up with * |
| 15:19.57 | [TK]D-Fender | https://www.google.ca/search?q=asterisk+cisco+7961g+guide |
| 15:22.12 | *** part/#asterisk Haris (~haris@unaffiliated/haris) |
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| 15:53.06 | skyroveRR | Is the configuration for asterisk version 13.3.2 compatible for asterisk version 13.9.1? |
| 15:55.40 | [TK]D-Fender | 99.999% |
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| 15:58.25 | skyroveRR | [TK]D-Fender: Hmm. I'm getting these notices on startup: http://pktsurf.in/files/asterisk.runtime.log.txt ; also my 13.3.2 loaded the least amount of modules, but it seems this one loads ALL the modules that are there. |
| 15:59.31 | *** join/#asterisk Nugget (~nugget@rennsport.macnugget.org) |
| 15:59.58 | skyroveRR | [TK]D-Fender: may you please have a look? |
| 16:01.59 | [TK]D-Fender | Why are you starting * as ROOT? |
| 16:02.30 | [TK]D-Fender | You're also only running at verbose 1 who is low |
| 16:02.32 | skyroveRR | Why shouldn't I? It complains about "permission denied" errors.. |
| 16:02.45 | [TK]D-Fender | and we aren't looking at your configs folder here when we should be |
| 16:02.49 | [TK]D-Fender | esp as it says some are missing |
| 16:03.02 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 16:03.06 | [TK]D-Fender | You should not be running * as root |
| 16:04.18 | skyroveRR | Uh, ok. I'll fix that. :) |
| 16:06.42 | *** part/#asterisk luxio (~luxio@unaffiliated/luxio) |
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| 16:29.09 | kchehab | anyone try asterisk with linphone chat jaber Xmpp |
| 16:31.52 | wyoung | hey gang |
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| 17:29.29 | salz212 | Hi guys, I was wondering if we can use extension-states with hints mapped to and externally registered SIP device. So instead of asterisk own device we map to some other SIP endpoint e.g. 100@NON-Asterisk-IP |
| 17:31.09 | [TK]D-Fender | And what is managing that end? |
| 17:31.50 | salz212 | just SIP registrations let say registration are on some other sip server not asterisk. |
| 17:36.55 | salz212 | does that make sense? |
| 17:46.55 | [TK]D-Fender | * doesn't just do distributed service to other random servers. |
| 17:47.08 | [TK]D-Fender | It CAN do to another * via XMPP, but that's not "phone-based:" |
| 17:49.29 | salz212 | right but the subscription of device states is based on SIP so instead of send PUB/SUB/NOTify to asterisk can it not send it to sip/{EXTEN}@SIP-Server ? |
| 17:50.27 | salz212 | so basically what I am trying to achieve is have sip registerations on SIP server like SER and use asterisk queues scaling to multiple asterisk servers. |
| 17:54.00 | *** join/#asterisk SunGod (~SunGod@92.55.116.125) |
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| 18:02.01 | scv | on the topic of shared state to another asterisk server |
| 18:02.05 | scv | is xmpp the only option still |
| 18:03.24 | *** join/#asterisk K0HAX (~michael@shellhost.home.englehorn.com) |
| 18:05.07 | file | Asterisk can exchange device and MWI state using XMPP, SIP, or Corosync |
| 18:05.21 | file | in 14 it can also publish extension state to another SIP server using PUBLISH |
| 18:05.30 | file | it does not allow monitoring of stuff elsewhere |
| 18:08.04 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
| 18:25.19 | *** join/#asterisk BeachBall (~Death_BuB@unaffiliated/beachball) |
| 18:25.35 | BeachBall | can i get help with asteriskNOW in here? |
| 18:25.41 | BeachBall | or do they have their own chan? |
| 18:26.04 | WIMPy | Not for the GUI. Otherwise |
| 18:26.06 | WIMPy | ~ask |
| 18:26.06 | 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. |
| 18:26.42 | BeachBall | My problem is I had to rebuild an asterisk system from 2011 - I installed asterisknow - and can't find where to put my context's in |
| 18:27.07 | BeachBall | it's a very basic config, should I just install as normal and edit the files ? |
| 18:27.35 | WIMPy | The basic structure hasn't changed. |
| 18:28.23 | WIMPy | It depends if you installed with or without FreePBX. If you installed that as well, then hand editing is rather special. |
| 18:28.28 | salz212 | @file: thanks for your valuable input. By not allowing monitoring of stuff else where may I conclude following use-case not possible => an agent in an asterisk queue is not registered on asterisk server itself instead it is registered on another SIP server say kamailio/opensips would asterisk not load-balance queue calls to them based on some presence? |
| 18:29.13 | file | Asterisk would not have any knowledge of it, it would assume it is not in use and would only go in use on the server it is called on |
| 18:32.32 | [TK]D-Fender | Depends what GUI it's running if any |
| 18:33.04 | salz212 | @file: so in that case if we have only 1 such agent in queue, asterisk queue will keep on handing calls to the same external agent not knowing it's state, correct ? or would asterisk queue internals would keep a channel of such external call and won't send it more calls? |
| 18:33.26 | file | that _Asterisk_ wouldn't send calls if it is in use |
| 18:38.09 | salz212 | @file: great so that Queue-Asterisk server will keep state of it's calls / queue to make that decision. Considering that if I have another Queue-Asterisk-2 where I want to force-set such state lets say in-use that queue would also not send call to that external extension. |
| 18:38.39 | salz212 | @file: But since it is not a registration or an extension how can I set it's state on Queue-Asterisk-2 |
| 18:38.54 | WIMPy | Well, the queue does not keep state. The channel dialled, might. |
| 18:40.20 | file | the queue keeps some state - it knows if a channel is in use/not in use and if it's dialed and such |
| 18:40.35 | file | exchanging that needed information is either done like I mentioned with XMPP/PJSIP/etc, or up to you to do somehow |
| 18:40.51 | WIMPy | Err. That's not what I found so far. |
| 18:42.00 | file | it does, if stuff is configured right and device state is such that the state has changed |
| 18:42.13 | file | and recent versions do keep track of whether a channel is dialed or not to stop other queues |
| 18:42.32 | WIMPy | That's what I say. It's device state, if available. |
| 18:43.50 | salz212 | @file: right, I did that with XMPP earlier using tigas as a presence server it was okay but in that scenario I was distributing device states of sip end-point registered at Asterisk. And queues worked fine. But now I am discussing the possibility of doing the same but with external SIP registrations at kamailio/opensips, what would it be distributing in this case as we do not have devices attached. |
| 18:44.10 | file | you would need to have devices in Asterisk as well |
| 18:44.21 | file | if not, then you would have to design something yourself |
| 18:44.31 | file | perhaps using custom device states or something else |
| 18:46.31 | salz212 | @file: sure, just brainstorming to get ideas and getting validated. |
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| 21:34.45 | apb1963_ | The install_prereq script in ../contrib/scripts uses awk to select the line beginning with "p" in ubuntu. p stands for purged. Can someone tell me why it seeks to install purged packages? The only reason I really care is because it aborts and I'm unable to resolve the issue. |
| 21:35.59 | apb1963_ | That is, other than removing it as a pre-req, or selecting a different package manager and modifying the script in a variety of ways. |
| 21:43.57 | apb1963_ | I found this note on the wiki:You should always use your operating system's package management tools to ensure that your system is running the latest software before running install_prereq. Ubuntu 14's libsnmp-dev package, for instance, has an issue where it will attempt to remove critical system packages if the system isn't updated before an attempt is made to install that package |
| 21:44.09 | apb1963_ | That happens to be the exact package in question. |
| 21:44.32 | apb1963_ | And yes... everything was up to date |
| 21:45.02 | apb1963_ | Which is why I ask the question about purge. If the script looked for "i", it would know that the package was already installed. |
| 21:46.22 | apb1963_ | well... it's actually trying to resolve between: |
| 21:46.24 | apb1963_ | i libsnmp-dev |
| 21:46.25 | apb1963_ | p libsnmp-dev:i386 |
| 21:47.06 | apb1963_ | So, it's perplexing why a purged package would be favored over an installed one. |
| 22:42.04 | lanning | x86_64 vs. i386? |
| 22:49.09 | apb1963_ | yes |
| 22:49.18 | apb1963_ | it would seem |
| 22:51.13 | apb1963_ | lanning: yes...but I've just done discussing with the ubuntu guys and they think the script is "terrible". My own opinion is someone didn't test... or understand exactly what they were doing. It's not terrible... but does need some rewriting. |
| 22:52.14 | lanning | :) |
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| 23:14.32 | scv | apb1963_: its in contrib |
| 23:14.41 | scv | so, not official or supported |
| 23:16.17 | apb1963_ | it would be nice if people tested it before distributing it as if it worked. |
| 23:16.24 | apb1963_ | My system is now hosed. |
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