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| 00:29.46 | darkunderlord | ok, so I think it's been happening since I was trying to get upgraded to 13. I'm guessing it's DB changes I made to go to 13, and 11 doesn't like them. Odd though, because it seems to have all the data it needs when querying from the CLI. |
| 00:31.08 | Samot | Well there were some changes. |
| 00:31.15 | Samot | Outline in the wiki. |
| 00:31.18 | Samot | Well mainly 11 to 12 |
| 00:31.24 | Samot | LTS never has new stuff in it |
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| 08:37.57 | zicada | ls |
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| 10:51.25 | zicada | I'm having some issues with 2 eventfd file discriptors sticking around after each successful call. Thus after a while, the underlying OS will complain about too many open files. |
| 10:51.58 | zicada | This is on a ubuntu lts. |
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| 11:21.05 | zicada | Are any of the developers of Asterisk in this channel ? |
| 11:21.16 | Samot | What is your issue? |
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| 12:27.09 | blinky_ | afternoon all. I am running an asterisk box with multiple ring groups. I need to set a time condition for between 8 and 9 in the morning. The rule needs to divert all ring groups to an other ring group. I can set up the time condition but I dont understand where to set the ring groups that need to be diverted, any advice? |
| 12:28.01 | Samot | Uhm. |
| 12:28.10 | Samot | You set them up like normal and route calls to them |
| 12:37.44 | *** join/#asterisk FarhaadN (~Farhad@82.99.206.194) |
| 12:38.27 | FarhaadN | hi, how can i set two ip address for bind address in sip.conf |
| 12:38.47 | FarhaadN | i want to bind to internal and external ip address |
| 12:38.47 | Samot | You don't |
| 12:38.59 | Samot | Unless what is v4 and one is v6 |
| 12:39.23 | FarhaadN | why i can set 0.0.0.0 |
| 12:39.28 | FarhaadN | but i cant 2 address? |
| 12:39.35 | Samot | That's how it is |
| 12:39.43 | Samot | That's how it has always been with Chan_SIP |
| 12:40.05 | FarhaadN | only way is set bind address to 0.0.0.0? |
| 12:40.28 | Samot | You can bind to 0.0.0.0, that's the wildcard |
| 12:40.36 | Samot | Or you can put the specific address in there |
| 12:40.45 | roswell | FarhaadN, afair chan_pjsip has an opportunity to point several transports (or whatever it's called) to different ips, however i'm not a big fan of it |
| 12:40.51 | Samot | But you cannot have two IPs listening on UDP for Chan_SIP. |
| 12:41.05 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 12:41.16 | Samot | You can have two listen, one for UDP and once for TCP |
| 12:41.29 | Samot | But not multiple IPs on the same transport. |
| 12:41.50 | roswell | FarhaadN, what's your asterisk version? |
| 12:42.06 | FarhaadN | Samot: OK |
| 12:42.11 | FarhaadN | roswell: 14.6 |
| 12:42.53 | Samot | So this box has both a public IP and a LAN IP? |
| 12:43.03 | Samot | That you want to listen for SIP? |
| 12:43.13 | roswell | FarhaadN, have a look at https://wiki.asterisk.org/wiki/display/AST/PJSIP+Transport+Selection , those sections having type=transport |
| 12:43.32 | FarhaadN | i have to interfaces |
| 12:43.38 | Samot | OK |
| 12:43.47 | FarhaadN | one for internal : 192.168.2.2 |
| 12:43.58 | FarhaadN | and onw for register sip frome internet |
| 12:44.03 | Samot | So this box isn't behind NAT? |
| 12:44.20 | Samot | Is it behind a firewall at least? |
| 12:44.55 | FarhaadN | yes |
| 12:45.35 | FarhaadN | roswell: check it, both use tcp |
| 12:46.10 | Samot | You'll need to switch to PJSIP for this. |
| 12:46.32 | Samot | Although, I honestly can't say why you are doing it this way. |
| 12:46.48 | Samot | "I have two interface cards" is never enough reasoning. |
| 12:47.30 | FarhaadN | tell me how figure that |
| 12:47.39 | Samot | Figure what? |
| 12:48.17 | roswell | FarhaadN, check what? see that bind keyword? |
| 12:48.47 | FarhaadN | Samot: tell me how to fix this |
| 12:48.54 | Samot | Fix what? |
| 12:48.59 | Samot | What is actually broken? |
| 12:49.37 | FarhaadN | i want to other people from internet register to my asterisk server with internet ip address |
| 12:49.56 | Samot | Uhm. |
| 12:50.01 | Samot | That's called NAT |
| 12:52.25 | FarhaadN | Samot: ask from network manager to nat that ip too local ip? |
| 12:53.50 | Samot | FarhaadN: Of course. |
| 12:54.05 | FarhaadN | i dont need change anything? |
| 12:54.18 | thiagoc | hi all, I'm having a problem when some extension pick up a call from a queue and transfer to another extension, asterisk creates a local channel and when the call terminates that local channel doesn't disappears |
| 12:54.21 | Samot | FarhaadN: Are you telling that after all this time coming in here, you've never set this part up? |
| 12:54.28 | Samot | FarhaadN: Of course. |
| 12:54.37 | Samot | You need to read the wiki and the sample configs for this |
| 12:54.57 | FarhaadN | Samot: i only set 0.0.0.0 |
| 12:55.07 | Samot | I'm not talking about that |
| 12:55.10 | Samot | I'm talking about NAT |
| 12:55.18 | FarhaadN | no, i mean in past |
| 12:55.20 | Samot | Yes, you setup NAT to send the calls to the PBX |
| 12:55.33 | Samot | I'm talking about setting up Asterisk. |
| 12:57.21 | FarhaadN | Samot: can u give me link of wiki? |
| 12:58.16 | pawiecki_ | FarhaadN: https://wiki.asterisk.org/wiki/display/AST/Home |
| 12:58.59 | pawiecki_ | also https://www.digium.com/training/asterisk-books |
| 13:02.27 | FarhaadN | pawiecki_: tanx |
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| 13:32.46 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:44.39 | zicada | I tried asking about this earlier, but people were likely asleep. |
| 13:45.43 | zicada | I am looking to solve a problem with leaking file descriptors. Every time we complete a call, two "zomby" eventfd descriptors are created and they are never removed, so after X calls we will hit the soft limit and Asterisk crashes with "too many open files" |
| 13:46.14 | zicada | This occurs when we use Asterisk to bridge together two channels mixing in audio. |
| 13:47.08 | zicada | I have been peering at the code,and google'd a lot, but no luck. It very well could be we are doing something "weird" exposing a cornercase where this happens, but no sure. |
| 13:47.23 | roswell | zicada, how exactly do you bridge channels? |
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| 13:52.54 | zicada | roswell: we are bridging together an incoming 1st leg and and outgoing second leg (like a conferance). |
| 13:53.23 | roswell | zicada, do u use any external app for mixing streams? |
| 13:55.32 | zicada | roswell: no |
| 13:56.05 | roswell | zicada, well without seeing your dialplan related to the issue, i can't tell more |
| 13:56.47 | roswell | usually asterisk is running out of fds when an external process fails to terminate properly |
| 13:57.07 | roswell | maybe there's some agi call |
| 13:57.47 | roswell | maybe some shell command |
| 13:59.29 | zicada | We are using ARI4Java to communicate with our apps |
| 14:00.05 | zicada | But shouldn't asterisk be in charge of closing down its own descriptors when communication no longer happens over its websocket(s) ? |
| 14:02.23 | roswell | zicada, do you have a chance of running netstat -apn | grep 8088 |
| 14:02.55 | roswell | or whichever port your asterisk uses for webrtc connections. just don't paste the whole output here |
| 14:03.08 | zicada | tcp 0 0 0.0.0.0:8088 0.0.0.0:* LISTEN - |
| 14:03.18 | zicada | and then 11 or so TIME_WAIT and one ESTABLISHED |
| 14:07.59 | roswell | so this is not about networking at least |
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| 14:10.15 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:11.16 | zicada | roswell: if we dont "addchannel" for the conferance, we only end up with 1 descriptor "left over" instead of 2. |
| 14:11.40 | zicada | And the way we trigger it is simply making a call and then hanging up after a few seconds. |
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| 14:46.17 | rrittgarn | having an issue with app_queue. Periodic announcements aren't playing at all. Config is here: https://pastebin.com/14wW6APJ |
| 14:46.47 | rrittgarn | curious if I'm missing something obvious. I also don't see any attempts to play the files in debug output when logged, so guessing i missed something |
| 14:50.51 | rrittgarn | nevermind, figured it out, announcements wont play while ringing, so timeout has to be < my announce interval... now to figure out why its not playing my particular audio |
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| 15:28.06 | [TK]D-Fender | timeout=100 |
| 15:28.11 | [TK]D-Fender | <PROTECTED> |
| 15:28.18 | [TK]D-Fender | Yup, you scrweed up |
| 15:28.32 | [TK]D-Fender | it dials agents for 100 seconds before having a chance to play announcements |
| 15:28.41 | [TK]D-Fender | but you set the LIFE of the caller in queue to 100 seconds |
| 15:28.45 | [TK]D-Fender | so they never hear it |
| 15:29.10 | [TK]D-Fender | it doesn't play messages WHILE ringing agents, only in between |
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| 15:57.33 | *** mode/#asterisk [+o rmudgett] by ChanServ |
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| 16:19.54 | salviadud | I'm pondering if there's an easier way to handle this next scenario |
| 16:20.16 | salviadud | Today, one of my sales agents got 2 calls on her sip phone, and wanted to add them to a conference. |
| 16:20.43 | salviadud | So, I helped out and just transferred both parties to a conference room and dialed in to that room at the end. |
| 16:20.49 | salviadud | Is there an easier way? |
| 16:21.47 | salviadud | Maybe I am doing the best way there is. |
| 16:21.58 | salviadud | Or that would depend on what the phone can do. |
| 16:22.06 | salviadud | In my case, crappy cisco phones |
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| 16:26.34 | *** join/#asterisk miralin (~Thunderbi@91.237.94.4) |
| 16:38.02 | [TK]D-Fender | no "join" feature on those huh? |
| 16:38.17 | [TK]D-Fender | Polycom has that going back 12 years now.... |
| 16:41.19 | salviadud | There is a "join" softkey as I read a cisco manual, but I didn't find it available. |
| 16:41.59 | salviadud | But that might be with the SCCP protocol |
| 16:42.08 | salviadud | and I'm on sip right now... |
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| 18:14.13 | dan_j | Has anyone got experience disabling sipalg on a Technicolor TG589vac v2? |
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| 18:31.40 | Samot | Never even heard of them |
| 18:33.33 | salviadud | I did it! |
| 18:33.44 | salviadud | Those old cisco phones still work |
| 18:33.49 | salviadud | yeepee |
| 19:10.30 | *** part/#asterisk catbehemoth (~vasyl@lnglpq6100w-lp130-02-76-65-87-183.dsl.bell.ca) |
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| 19:25.28 | Ussat | afternoon, installing asterisk as a POC , I have a couple choices of Linux distros, RHEL 6 or 7, Ubuntu 16.04 LTS, are their packages for either of these ? |
| 19:26.01 | Ussat | I would prefer NOT to compile from source if can be avoided |
| 19:29.25 | allenp | I'm on ubuntu 17.04 and the packages are available there |
| 19:29.35 | allenp | I would assume anything on 16.04 would be out of date |
| 19:29.42 | allenp | or, older at least |
| 19:30.07 | [TK]D-Fender | it should have * 13 available |
| 19:30.08 | Samot | Just note that packages are always going to be out of date. |
| 19:30.36 | [TK]D-Fender | RH based you can use repos like the ones provided by FreePBX for just *'s basics |
| 19:30.47 | [TK]D-Fender | Not sure if Digium offers one still |
| 19:31.07 | Ussat | allenp, yea, we would like to use a LTS version of Ubuntu though |
| 19:31.18 | kraylo | Can you believe that an SMB that I consult for is actually considering Trixbox as an alternative to SwitchVox!? wtaf!!? |
| 19:31.21 | Samot | 16.04 |
| 19:31.34 | Samot | Trixbox == WAY DEAD |
| 19:31.44 | kraylo | yup! tried telling them that |
| 19:31.45 | Ussat | Yea we are going with 16.04 LTS |
| 19:31.46 | Samot | Not even like, "It hasn't been updated in a year" |
| 19:31.53 | Samot | More like IT DOESN'T EXIST |
| 19:31.55 | kraylo | their head office still uses it; so it's on the table |
| 19:31.59 | Samot | No. |
| 19:32.02 | Samot | tell them no. |
| 19:32.12 | Samot | That's like saying "I still run Windows XP so that's an option" |
| 19:32.21 | kraylo | LOL! Yup. |
| 19:32.34 | Samot | It will not support anything current |
| 19:32.37 | Samot | It will not be supported |
| 19:33.01 | Ussat | allenp, did you run intp this bug with it on 17.04 ? https://bugs.launchpad.net/ubuntu/+source/asterisk/+bug/1458323 |
| 19:33.04 | Samot | So if you guys are cool with a 100% unsupport/unmaintained/out-of-date PBX system. Go for it. |
| 19:33.15 | [TK]D-Fender | DEAD FOR SEVEN FUCKING YEARS WITH OUTDATED FUNCTIONALITY, NO NEWER HARDWARE COMPATIBILITY ANCIENT SECURITY HOLES, NO SUPPORT, AND TONS OF UNRESOLVED BUGS. |
| 19:33.25 | Samot | ^^^ |
| 19:33.28 | [TK]D-Fender | If you pick TrixBox you're BEYOND stupid |
| 19:33.31 | Samot | But hey, the boss still runs it. |
| 19:33.48 | [TK]D-Fender | That's choosing to deliberately stab yourself in the face |
| 19:33.54 | kraylo | problem is too many big mouths -- that aren't doing any of the actual work -- keep bringing it up. I've already told them.. it's their choice and that I will not be held responsible for the 'result' |
| 19:34.00 | Samot | And shoot yourself in the foot |
| 19:34.10 | Samot | Then set yourself on fire. |
| 19:34.17 | Ussat | [TK]D-Fender, I am not familiar with FreePBX, is that Asterisk ? |
| 19:34.17 | [TK]D-Fender | I think the head-stabbing is a higher priority than your foot :) |
| 19:34.19 | kraylo | more like lobbing off your own d*ck |
| 19:34.20 | Samot | With a tire around your neck |
| 19:34.30 | Samot | No. |
| 19:34.34 | Samot | It's all of the above. |
| 19:34.34 | [TK]D-Fender | Trixbox was a shitty FORK of FreePBX |
| 19:34.35 | kraylo | anyway... just had to vent |
| 19:34.43 | [TK]D-Fender | both of which use * underneath |
| 19:34.44 | allenp | Ussat: I have not run into any issues with my install, however I'm not running anything production at this point. Just a lab setup to learn asterisk |
| 19:34.44 | Samot | I wouldn't recommend it. |
| 19:34.59 | Samot | kraylo: I'm giving you professional advice. Do not use Trixbox. |
| 19:35.06 | Ussat | allenp, yea this is just a POC |
| 19:35.10 | [TK]D-Fender | 7 year old shit |
| 19:35.16 | [TK]D-Fender | fuck POC |
| 19:35.20 | [TK]D-Fender | there's no excuse for it |
| 19:35.28 | kraylo | Samot: I'm 100% in agreement. I will not use or implement it. |
| 19:35.28 | Ussat | allenp, you just did a apt-get install asterisk , correct ? |
| 19:35.36 | [TK]D-Fender | You want to install Windows 3.11 while you're at it? |
| 19:35.43 | allenp | Ussat: yeah |
| 19:35.49 | Ussat | OK |
| 19:35.51 | kraylo | Samot: if 'they' choose to, despite my recommendation to the contrary... that's their problem. |
| 19:35.58 | [TK]D-Fender | Also.. you POC will not have the modern interfaces on * as it is. |
| 19:35.59 | Ussat | what version of asterisk did you end up with ? |
| 19:36.04 | allenp | you can always apt-cache search to see what is available |
| 19:36.32 | Ussat | true |
| 19:36.34 | allenp | no idea, just installed it over the long weekend. I'd check but this is on a VM on my home network |
| 19:36.48 | Ussat | ahh, yea this is for a VM also |
| 19:36.51 | Ussat | OK, thanks |
| 19:37.09 | allenp | no prob |
| 19:38.42 | Samot | kraylo: Are you the one that has to deal with it? |
| 19:38.47 | Samot | Because if you are, it's your problem. |
| 19:39.01 | Samot | Or if you're in charge of IT/tech in some way. 100% your issue. |
| 19:41.05 | *** part/#asterisk drudge` (~drudge@unaffiliated/drudge/x-837452) |
| 19:43.42 | Ussat | allenp, you grabbed that from universe right ? |
| 19:43.58 | allenp | good question |
| 19:44.09 | allenp | in the lab I just enable everything and go crazy |
| 19:44.24 | Ussat | heh ok |
| 19:44.28 | allenp | enable everything but backports |
| 19:45.32 | Ussat | Yea I am building this POC on my home laptop in a VM, assuming this does what I want, will export the OVA to our esxi cluster for a POC there |
| 19:46.45 | [TK]D-Fender | What are you trying to "prove"? |
| 19:48.12 | Ussat | short version, build a replacement for the old crap asterisk installs we have, running on CentOS5 with a gods know how old version of asterisk |
| 19:48.40 | Ussat | its not a full blown PBX by any means, its used for a specific application |
| 19:50.09 | [TK]D-Fender | What is there to actually "prove" in going from a questionable origin * build to a know dead-end pit-stop from history? |
| 19:50.34 | [TK]D-Fender | because if you had "specific application" ..... Trixbox isn't going to offer you anything for this at all |
| 19:50.53 | [TK]D-Fender | It's a dumb shit GUI you'll still have to add all your custom shit to |
| 19:51.08 | Samot | I think you're confusing two conversations. |
| 19:51.12 | [TK]D-Fender | Which will prove nothing ... except that it installs * and you still ahve to adapt your custom stuff to it |
| 19:51.16 | Samot | kraylo = Trixbox |
| 19:51.17 | [TK]D-Fender | oops |
| 19:51.21 | [TK]D-Fender | yeah... |
| 19:51.24 | Samot | Ussat = Asterisk upgrade |
| 19:51.41 | [TK]D-Fender | devloped a nifty headache and the ibuprofen hasn't kicked in yet... |
| 19:51.42 | Samot | Although.. |
| 19:51.49 | [TK]D-Fender | scratch that |
| 19:51.56 | Samot | I'm not sure what Proof of Concept is needed for this |
| 19:52.00 | [TK]D-Fender | Don't mind me... |
| 19:52.10 | Samot | Can you do it on the current version of Asterisk you're running? |
| 19:52.16 | [TK]D-Fender | You can be unsure of his need... I am not entirely here right now. |
| 19:52.22 | [TK]D-Fender | goes for a walk to get some air |
| 19:52.27 | Samot | If that's a yes, 99.999% chance it's going to run on the upgraded box |
| 19:53.02 | Ussat | Samot, its complicated, mostly political |
| 19:53.17 | Samot | OK |
| 19:53.19 | Samot | So. |
| 19:53.24 | Samot | You need a new OS |
| 19:53.30 | Ussat | and NO, I really cant, problem is mainly, its on a Centos5 system, old as hell, getting no updates etc |
| 19:53.35 | Ussat | Samot, exactle |
| 19:53.36 | Samot | But want a LTS... |
| 19:53.46 | Samot | But then.. |
| 19:53.50 | Samot | You want to install from packages.. |
| 19:53.55 | Samot | Which are always behind. |
| 19:54.15 | Samot | The Asterisk packages are always a couple releases behind. |
| 19:55.05 | Samot | Or and not maintained by Asterisk/Digium. |
| 19:55.13 | Samot | So what they do to them to make them work... |
| 19:55.31 | Samot | Can't always be know. |
| 19:55.40 | Ussat | Samot, I understand that, BUT, a LTS will get support/sec updates etc for a set ammount of time. I am OK with "older" packages, but not ancient. Also, I will be able to update the Asterisk install as new packages are released |
| 19:56.00 | [TK]D-Fender | Nobodyis making packages for C5. |
| 19:56.03 | Samot | Ussat |
| 19:56.04 | Ussat | Could I do a source install on RHEL7 or 17.04, sure |
| 19:56.12 | Samot | Let me give you a REAL example. |
| 19:56.15 | [TK]D-Fender | So forget LTS |
| 19:56.18 | [TK]D-Fender | that's just game over |
| 19:56.24 | Ussat | [TK]D-Fender,.... |
| 19:56.29 | Ussat | dude I am MOVING off that |
| 19:56.30 | Samot | He's talking about a new OS |
| 19:56.34 | [TK]D-Fender | ok |
| 19:56.51 | Samot | Not using Ubuntu 17 vs 16 because 16 is LTS and 17 is not |
| 19:57.00 | Samot | My point is.. |
| 19:57.06 | Samot | The Asterisk install using packages.. |
| 19:57.32 | Ussat | will always be behind.......I know |
| 19:57.32 | Samot | 6 days ago Digium just patched Asterisk 13, 14 AND 11 |
| 19:57.38 | Samot | 11~~~~ |
| 19:57.51 | Samot | It's 30 days from being fully dead |
| 19:57.57 | Samot | It's Security Fixes Only |
| 19:58.11 | Samot | The bug they patched is so bad and been around for so long.. |
| 19:58.14 | Samot | They had to patch 11 |
| 19:58.21 | Samot | Any package you install now... |
| 19:58.28 | Samot | Will still have that major security bug |
| 19:59.06 | Ussat | Samot, I understand what you are saying, but I am not going to be upgrading source based installs every time there is a patch |
| 19:59.16 | Samot | So note right now, anything version of Asterisk 13 that isn't 13.17.1 will have a major security bug |
| 19:59.27 | Samot | MAJOR SECURITY BUG |
| 19:59.37 | Samot | Those are the "Yes, we should update" |
| 20:00.31 | Samot | This isn't a patch that add Taylor Swift ring tones to Asterisk. |
| 20:00.40 | Ussat | ... |
| 20:00.42 | Samot | This is a patch that stops your ass from hanging out |
| 20:03.53 | Samot | I dont read PMs |
| 20:04.15 | Ussat | ok |
| 20:04.40 | Ussat | anyway..I get what youre saying, and I will look at the sec issue patched |
| 20:11.27 | kraylo | Samot: it won't be my problem (trixbox) since (1) technically, it's not my function to support it after its been implemented (2) I'll be making it clear that I do not offer support nor will I personally implement that outdated P.O.S.. I'd be happy to walk away from the whole deal rather than get snaked into such a situation. |
| 20:11.49 | Samot | Good call |
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| 21:29.14 | Bhakimi | hi guys, i randomly get these type of errors "Max retries exceeded to host xxx.xxx.xxx.xxx on IAX2/provider-trunk-2087813854-2986 (type = 6, subclass = 2, ts=357002, seqno=60) " |
| 21:29.38 | Bhakimi | eventually the channel driver locks and nothing works until i restart asterisk |
| 21:30.08 | Bhakimi | any idea how i can force the channel to drop or some setting i can possibly set to prevent this lock ? |
| 21:47.12 | Samot | That sounds like your SIP messages are timing out |
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| 21:47.38 | *** mode/#asterisk [+o gtjoseph_] by ChanServ |
| 21:49.07 | Samot | There aren't any real IAX2 commands for this but AMI or ARI should be able to destroy the channel. |
| 21:49.16 | Samot | But they'll need to know it exists first |
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| 23:59.22 | hdon | hi all :) anyone with experience using node-ari-client? the result of getChannelVar() seems to be a channel object rather than the value of the channel variable i asked for. am i using it wrong? |