| 00:05.30 | *** join/#asterisk Diogo (sid37244@gateway/web/irccloud.com/x-ejuxbwxulukqtgbf) |
| 00:17.18 | Navion | [TK]D-Fender: Do you know about "fail2ban"? |
| 01:00.48 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
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| 02:44.16 | jeffspeff | Navion, what's your question about f2b? |
| 02:53.22 | Penguin | I think he was just testing [tk]d-fender's knowledge. |
| 02:54.04 | *** join/#asterisk ThatDamnRanga (~wiretap@unaffiliated/wiretap) |
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| 03:22.47 | [TK]D-Fender | Navion: Yes |
| 03:41.01 | jpsharp | And it seems that all my analog dial problems went away by going from Asterisk 1.6.2-22 to 1.6.2-24 |
| 03:41.57 | [TK]D-Fender | sics a few weasels on jpsharp |
| 03:43.07 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 03:46.17 | jpsharp | I'd dial out on an analog line, the call would go out and get answered, but Asterisk would never recognize the DAHDI channel as having been answered....yet I had bidirectional audio. |
| 03:46.56 | jpsharp | And my IP phone was still saying "183 Session Progress". |
| 03:53.52 | [TK]D-Fender | call progress detections issues most likely |
| 03:54.02 | [TK]D-Fender | which is why the default is to assume an answer |
| 04:33.07 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 04:57.53 | jpsharp | But it never assumed an answer. the whole thing occurred while the outbound dahdi channel was still in a Dialing state. |
| 04:57.59 | jpsharp | So I was just really darn confused. |
| 04:58.13 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 05:00.45 | [TK]D-Fender | you should disable progress detection at that point |
| 05:01.25 | jpsharp | It was off. "callprogress=no" |
| 05:01.32 | jpsharp | "busydetect=no" |
| 05:02.27 | Kobaz | awesome=yes |
| 05:02.37 | jpsharp | Oh, that's what I forgot. |
| 05:02.41 | jpsharp | :) |
| 05:02.53 | jpsharp | And "doIwantittoworkright=yes" |
| 05:02.58 | Kobaz | dont dont forget to set nobugs=yes |
| 05:04.00 | Kobaz | jpsharp: "from Asterisk 1.6.2-22 to 1.6.2-24"... |
| 05:04.03 | Kobaz | that's interesting |
| 05:04.17 | Kobaz | because that's a distro version |
| 05:04.26 | jpsharp | asterisk 1.6.2.24 |
| 05:04.28 | Kobaz | which generally includes only very small modifications |
| 05:04.29 | Kobaz | oh |
| 05:04.36 | Kobaz | that's not then |
| 05:04.53 | Kobaz | big difference between - and . |
| 05:05.05 | Kobaz | aaaand why are you still on 1.6? |
| 05:05.21 | Kobaz | Navion: fail2ban is meh |
| 05:05.29 | Kobaz | Navion: it's failed on me a few times |
| 05:05.47 | Kobaz | let attackers continue to brute force when the log lines clearly match ban rules |
| 05:06.12 | Kobaz | next random when-i-get-the-time project is write my own |
| 05:06.18 | Kobaz | i just don't trust it |
| 05:06.37 | Kobaz | i think fail2ban is named correctly, haha |
| 05:23.03 | ChannelZ | Well there is a little lag from when it sees the log lines, can block in the firewall, and then depending on how fast they've barfed traffic at you, when the firewall actually starts dropping shit |
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| 05:46.14 | fling | Can you guys help me starting medooze mcu? http://dpaste.com/08EZ8RP Caused by: java.lang.IllegalArgumentException: JBWEB000250: Child container with name already exists |
| 05:53.40 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:da4:221:6aff:feb8:e0b2) |
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| 06:11.36 | jpsharp | Why still on 1.6? cause I don't feel like going through the clusterflop evolution required to install something more recent on a Centos 5.8 box. |
| 06:11.52 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 06:12.17 | jpsharp | And why Centos 5.8? Because the box is 900 miles away and doesn't really belong to me. I'm just the AstMonkey. |
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| 07:46.23 | dkakoti | Can anyone suggest how to create PRI line failover in dialplan? |
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| 08:23.23 | r00f | dkakoti, what do you mean by failover? |
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| 08:47.47 | dkakoti | @r00f For outbound call suppose I have one dial group to dial,e.g. DAHDI/g2/9233333 .If g2 fails calls are automatically redirected to another pri |
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| 09:09.01 | wasanzy | Hi folks |
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| 09:19.18 | phix | GANG! |
| 09:23.24 | r00f | dkakoti by "fails" you mean "no free channels" ? |
| 09:23.46 | wasanzy | am still finding a way to play audio file in asterisk directly from aurl |
| 09:24.08 | r00f | because if for example callee is busy or not available, no sense to try to call him again via another line |
| 09:24.13 | wasanzy | MP3Player and playback doesn't seem to work for me |
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| 10:54.01 | BlackDex | Hello there, i have some cisco phones, and i want to have the intercom pickedup via the speakers instead of the headset, but i can't seem to get this to work, someone a clue about this? It always plays via the headset instead of the speaker |
| 10:56.48 | phix | I can do that with snome' |
| 10:56.50 | phix | snom* |
| 10:56.56 | phix | never tried with cisco phones though |
| 11:00.05 | Milos | I broke something. |
| 11:01.11 | Milos | Wait, I didn't! |
| 11:01.12 | Milos | WTF |
| 11:05.31 | Milos | is there a command to force reregistration? |
| 11:05.36 | Milos | and/or deregistration? |
| 11:12.35 | *** join/#asterisk areski (~areski@188.188.42.107) |
| 11:21.05 | dkakoti | @r00f No callee is not busy, If your PRI outgoing line through which you are generating the call is DOWN? |
| 11:28.33 | Milos | I'm having some trouble understanding how to register another extension from the same provider. |
| 11:30.21 | *** join/#asterisk calum_ (~calum_@79-75-65-230.dynamic.dsl.as9105.com) |
| 11:30.24 | Milos | Is this correct? https://bpaste.net/show/b2eb54ee771c |
| 11:30.35 | Milos | I get "Forbidden" when I try dial out of line1 |
| 11:40.15 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 11:41.39 | Milos | okay, I changed it to type=friend and added fromuser and now it works... |
| 11:41.45 | Milos | I'll need to read into why |
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| 11:54.39 | *** join/#asterisk eduzimrs (~adada@mail.aytycrm.com.br) |
| 11:55.24 | eduzimrs | hi guys,is there a programing language the works best with AGI scripts? |
| 11:55.38 | eduzimrs | hi guys,is there a programing language that works best with AGI scripts? |
| 11:57.08 | eduzimrs | i mean, that is native with Asterisk |
| 11:57.19 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 12:00.31 | dkakoti | @eduzimrs PHP, phpagi library is available |
| 12:02.00 | eduzimrs | i've read about, and in some forums they say that PERL is a kind of builtin language that talks better to Ast |
| 12:04.05 | eduzimrs | if that is true i hope any dev could answer me |
| 12:07.01 | r00f | Asterisk::AGI cpan module acts just the same way as phpagi |
| 12:07.52 | r00f | im not developer of asterisk, but i've modified both libraries to my needs |
| 12:08.13 | r00f | and i've seen them from inside |
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| 12:29.19 | eduzimrs | understand |
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| 13:04.00 | *** join/#asterisk stizmatic21 (~steven@static-108-29-37-125.nycmny.fios.verizon.net) |
| 13:16.55 | stizmatic21 | Hi everyone. Im running into an issue with our PBX where SIP channels are not being destroyed. Running asterisk 1.8.7.0. At the end of the day the number of open SIP channels is around 2,000. The only way to get rid of them is to restart asterisk. I've noticed two odd log entries. One saying "Autodestricut on dialog ... with owner in place" and the other "sip_xmit of (random hex number) |
| 13:16.57 | stizmatic21 | to 0.0.1.181:5060 returned -1: Invlaid Argument". |
| 13:17.40 | stizmatic21 | Any idea where I should look? To add -- the PBX is NAT'd and I've checked all firewall/NAT rules to make sure the necessary ports are open. |
| 13:19.16 | *** join/#asterisk Distraught (~Distraugh@77-6-161.static.cyta.gr) |
| 13:19.21 | Distraught | aloha |
| 13:19.48 | Distraught | i have a problem with icalldroid device. I have cisco 7941G . I have 3 devices. I cant make outbound calls |
| 13:20.00 | Distraught | When i had one device i could make calls |
| 13:20.22 | Distraught | When i added a second one i cant even do an internal call |
| 13:20.44 | [TK]D-Fender | stizmatic21: You should look at upgrading your system. You are 25 releases behind and there have been tons fo fixes, quite possible some for whatever caused this. |
| 13:20.44 | Distraught | i get Using SIP RTP CoS mark 5 icalldroid |
| 13:20.47 | Distraught | in console |
| 13:21.08 | [TK]D-Fender | Distraught: "sip set debug on" <- |
| 13:21.14 | Distraught | ok |
| 13:21.21 | [TK]D-Fender | Distraught: the line you posted doesn't actually say anything useful. |
| 13:21.21 | Distraught | enabled |
| 13:21.29 | [TK]D-Fender | 'disNow go look at a call. |
| 13:21.37 | Distraught | <PROTECTED> |
| 13:21.45 | [TK]D-Fender | Also means nothing |
| 13:21.49 | Distraught | hmm |
| 13:22.39 | Distraught | anythng that could help ? |
| 13:23.25 | [TK]D-Fender | It's help to see actual debug in a pastebin. |
| 13:23.27 | [TK]D-Fender | ~pb |
| 13:23.27 | 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. |
| 13:23.30 | Distraught | can i pastebin? |
| 13:23.31 | Distraught | a ok |
| 13:23.33 | Distraught | one momment |
| 13:24.23 | Distraught | http://pastebin.com/VY0gKWmh |
| 13:25.20 | stizmatic21 | Thanks D-Fender, will look into that. |
| 13:26.16 | [TK]D-Fender | Distraught: That is just a tiny piece of the communication |
| 13:29.59 | *** join/#asterisk Distraught (~Distraugh@77-6-161.static.cyta.gr) |
| 13:30.05 | Distraught | sorry disconnected |
| 13:30.12 | Distraught | did you say anything Fender? |
| 13:31.36 | [TK]D-Fender | [09:26][TK]D-FenderDistraught: That is just a tiny piece of the communication |
| 13:31.59 | Distraught | i just called a number. Thats all that it outputs [TK]D-Fender |
| 13:32.15 | [TK]D-Fender | No, there are clearly TONS of lines before that point |
| 13:32.42 | Distraught | one moment i ll try again |
| 13:33.31 | Distraught | oh it does not copy everything from putty |
| 13:35.20 | Distraught | http://pastebin.com/cwhJjv2u [TK]D-Fender |
| 13:35.24 | Distraught | sorry for that |
| 13:38.33 | [TK]D-Fender | INVITE sip:6@192.168.1.254;user=phone SIP/2.0 Looking for 6 in DLPN_Viva (domain 192.168.1.254) SIP/2.0 484 Address Incomplete |
| 13:38.44 | [TK]D-Fender | You dialed "6" and there is no match for it. |
| 13:39.01 | Distraught | when i dial any number it shows that |
| 13:39.15 | Distraught | any number. I cant type the whole number |
| 13:39.16 | Distraught | it fails |
| 13:40.23 | [TK]D-Fender | That's an issue with your phone. |
| 13:40.51 | Distraught | but it called before |
| 13:40.57 | Distraught | i puted a second one |
| 13:43.20 | *** join/#asterisk theron (~theron@2620:10d:c0a4:1303:2acf:e9ff:fe1c:8fb) |
| 13:45.44 | [TK]D-Fender | You need to fix the dialplan on your phone |
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| 13:49.01 | *** join/#asterisk saint_ (~saint@ppp-198-15-108-194.phx.napinet.net) |
| 13:50.53 | Distraught | so it has nothing to do with dialplans of the asterisk box |
| 13:51.29 | [TK]D-Fender | Correct |
| 13:51.31 | [TK]D-Fender | fix your phone |
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| 13:58.54 | *** join/#asterisk saint_ (~saint@ppp-198-15-108-194.phx.napinet.net) |
| 13:59.51 | saint_ | hi all - any idea on how to backup the local database , so I can restore it in another version of asterisk ? |
| 14:01.21 | [TK]D-Fender | What database? |
| 14:01.43 | dkakoti | if mysql database the use mysql dump |
| 14:02.26 | saint_ | the one from within asterisk (console) |
| 14:02.30 | saint_ | database show |
| 14:02.39 | [TK]D-Fender | "core show settings" |
| 14:02.46 | [TK]D-Fender | AstDB" <- |
| 14:02.57 | [TK]D-Fender | at the bottom |
| 14:03.21 | saint_ | <PROTECTED> |
| 14:03.27 | saint_ | so i just need to copy this file ? |
| 14:03.30 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:03.52 | Penguin | Would you prefer a harder way? |
| 14:04.16 | saint_ | nah , just making sure. i m upgrading from 1.8 to 13. thanks for the tip. |
| 14:04.30 | Penguin | That might be a problem. |
| 14:04.34 | saint_ | why is that ? |
| 14:04.45 | [TK]D-Fender | Penguin: * should auto-migrate it. |
| 14:04.49 | [TK]D-Fender | Penguin: Shouldn't be a problem. |
| 14:05.00 | [TK]D-Fender | Penguin: the otehr way around... now that would have been a problem. |
| 14:05.09 | Penguin | Okay, good. |
| 14:05.22 | [TK]D-Fender | saint_: AstDB changed from BDB to SQLite in * 10 |
| 14:05.26 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 14:05.35 | [TK]D-Fender | saint_: * will auto-convert BDB if it finds it. |
| 14:05.46 | saint_ | [TK]D-Fender okay, thank you |
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| 14:28.05 | wasanzy | am getting this error when reloading cdr_custom module: |
| 14:28.08 | wasanzy | dr_custom.c:90 load_config: Unable to load cdr_custom.conf. Not logging custom CSV CDRs |
| 14:28.36 | wasanzy | don't know what to do, and google is not giving me results |
| 14:31.41 | *** join/#asterisk areski (~areski@188.188.42.107) |
| 14:34.04 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:34.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:43.53 | [TK]D-Fender | [10:28]wasanzydon't know what to do, and google is not giving me results <- have you considered looking for or at that file? |
| 14:44.28 | wasanzy | I have looked at the file |
| 14:44.35 | file | don't look at me! |
| 14:45.13 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:46.13 | wasanzy | hahah |
| 14:46.24 | wasanzy | I have reloaded it now. |
| 14:49.02 | wasanzy | do I need to add some line of code to extension.conf before my cdr_custom is logged, after reloading the module? |
| 14:49.04 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:49.57 | [TK]D-Fender | no |
| 14:50.51 | wasanzy | Master.csv have not been created after restarting asterisk |
| 14:53.25 | [TK]D-Fender | wasanzy: Nothing you've shown tells me that one should be created or updated... |
| 14:56.38 | wasanzy | http://pastebin.com/SPEuqq3b this are my configs |
| 14:57.39 | [TK]D-Fender | Where do we see taht the module is loaded and that your back-end shows up like it's supposed to? |
| 15:01.02 | wasanzy | http://pastebin.com/bFqKNAGS |
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| 15:05.02 | [TK]D-Fender | wasanzy: taht is not a CDR status dump |
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| 15:11.02 | wasanzy | http://pastebin.com/XJhyPeCd |
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| 15:11.59 | [TK]D-Fender | wasanzy: So show us a call and show us a folder dump |
| 15:13.50 | wasanzy | http://pastebin.com/jJLDMAp8 |
| 15:15.00 | wasanzy | hmmm is logging now |
| 15:15.31 | [TK]D-Fender | And I see some very clear errors being reported about what you configured |
| 15:17.01 | wasanzy | pbx.c:4621 pbx_extension_helper: No application 'ConfBridge' for extension (myconference, 602, 3)? |
| 15:17.21 | [TK]D-Fender | Does that say CDR? |
| 15:17.31 | [TK]D-Fender | (aside from all the other errors) |
| 15:17.48 | [TK]D-Fender | [2014-10-28 15:08:28.815] WARNING[7990][C-00000002]: pbx.c:4040 func_args: Can't find trailing parenthesis for function 'CSV_QUOTE(269.073015'? <----------------------------------- |
| 15:19.07 | wasanzy | [TK]D-Fender: yea but I have no idea the parenthesis it is talking about |
| 15:20.01 | [TK]D-Fender | It's telling you right to your face that you are missing a parenthisis there.... go look where you did it wrong in your config |
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| 15:22.57 | wasanzy | found it |
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| 16:11.58 | szonek | hi |
| 16:15.22 | szonek | i'm looking for a device to allow receiving voice calls (GSM) and at the same time forwarding caller ID information to the server (e.g. through USB, Ethernet, etc), any recommendations? |
| 16:15.45 | szonek | not sure how should i approach this |
| 16:19.35 | WIMPy | A USB stick or any phone with a USB or serial interface. |
| 16:20.11 | szonek | WIMPy: if USB stick how should i take calls? |
| 16:20.50 | [TK]D-Fender | szonek: Taking calls from a GSM interface is one thing.... what you want to TALK to them using is another |
| 16:20.55 | [TK]D-Fender | szonek: What do YOU want to use? |
| 16:21.05 | [TK]D-Fender | Asterisk works with all sorts of different things |
| 16:21.40 | [TK]D-Fender | VoIP phones (dedicated hardware device), using a variety of protocols (SIP, IAX2, SCCP, etc). |
| 16:21.54 | [TK]D-Fender | A soft-phone (same as above, but an app for a computer device). |
| 16:22.13 | [TK]D-Fender | a tdm analog phone (connected via an interface card that Asterisk supports) |
| 16:22.15 | [TK]D-Fender | etc |
| 16:22.19 | [TK]D-Fender | Take your pick |
| 16:22.26 | szonek | [TK]D-Fender: well i guess something like stationary phone would be alright, SIP probably? |
| 16:22.31 | [TK]D-Fender | But this is 100% separate from the device that is your GSM interface |
| 16:22.44 | [TK]D-Fender | A SIP phone would work just fine with Asterisk |
| 16:22.48 | [TK]D-Fender | (normally) |
| 16:23.02 | szonek | the simpler the better for me |
| 16:23.59 | szonek | if there would be something like stationary phone working with GSM and forwarding caller id info via some interface that would be perfect |
| 16:24.09 | szonek | but i'm not sure something like that exists :D |
| 16:24.32 | WIMPy | szonek: chan_dongle |
| 16:25.20 | WIMPy | Dou you onlt want to receive a caller ID or the actual call? |
| 16:26.10 | szonek | there are only two features that i need, receive a caller ID and send SMS |
| 16:26.31 | szonek | and of course receive call on the actual device |
| 16:26.43 | szonek | don't need to forward calls anywhere |
| 16:30.20 | WIMPy | Yes and no? |
| 16:30.55 | szonek | :D |
| 16:31.00 | szonek | sorry about that |
| 16:31.11 | [TK]D-Fender | [12:23]szonekASTERISK would be the software receiving the CallerID. |
| 16:31.39 | [TK]D-Fender | szonek: No gateway device is going to have an extra port on it just to shove out caller ID> This isn't a modem from 1985 we're talking about. |
| 16:31.44 | WIMPy | If oyu want to talk, use a USB donglw with chan_mobile, a GSM PCI card or an old Motorola phone with osmocom. |
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| 16:34.35 | WIMPy | There are also SIP GSM gateways, but make sure they can do SMS in a way you want. |
| 16:35.07 | szonek | WIMPy: let's say i use USB dongle, Asterisk can read the caller id and forward the call to the connected device (with earpiece and mic)? |
| 16:35.34 | szonek | that's how it works? |
| 16:35.37 | WIMPy | yes |
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| 16:38.03 | szonek | WIMPy: ok, i want to make it as simple as possible, could you recommend me any device to actually take calls? what protocol should i use? |
| 16:38.29 | dagelf | Where is the correct place to responsibly report privilege escalation exploits? |
| 16:39.45 | sgriepentrog | dagelf: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Vulnerabilities |
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| 16:40.01 | WIMPy | http://wiki.e1550.mobi/ |
| 16:40.03 | dagelf | thanks |
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| 16:41.37 | szonek | WIMPy: i mean the IP phone |
| 16:41.41 | szonek | WIMPy: any would work? |
| 16:43.32 | WIMPy | Whatever Asterisk supports. Which is quite a lot. |
| 16:45.51 | WIMPy | And why not just use a GSM phone? |
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| 16:50.04 | szonek | WIMPy: i have nothing against it, just thinking how to do it |
| 16:50.50 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 16:51.09 | szonek | WIMPy: i guess any old nokia phone would work? ;] |
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| 17:01.04 | szonek | sry, im back |
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| 17:22.17 | skrusty | anyone about with some fastagi knowledge? |
| 17:28.57 | Cuzner | it's a trap! |
| 17:32.01 | skrusty | :/ |
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| 18:30.13 | WIMPy | wonders why he sees so many calls with both billsec and duration =0. Is that a known result of something strange happening? |
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| 19:06.23 | apb1963 | what are people doing for billing on centos systems and * 1.8 ? |
| 19:06.50 | apb1963 | a2billing seems geared towards debian type systems |
| 19:08.13 | apb1963 | as per its own docs |
| 19:08.51 | [TK]D-Fender | Doesn't really matter |
| 19:10.06 | jpsharp | I've got a2b on assorted Centos and Debian/ubuntu boxen. It's not really distro-picky. |
| 19:10.15 | jpsharp | Just needs a reasonable installation of php & mysql. |
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| 19:45.19 | apb1963 | ok thank you |
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| 20:37.28 | eppigy | hello |
| 20:37.31 | eppigy | i am dave |
| 20:44.10 | pabelanger | hello yes, dog here |
| 20:46.09 | Cuzner | pabelanger! |
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| 21:02.01 | eppigy | hello dog I am dave |
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| 22:26.43 | dan_j | Hi. During a Dial, is it possible for the caller to enter a DTMF to move to another extension? |
| 22:26.54 | dan_j | Like you can do with a queue |
| 22:27.27 | WIMPy | core show application dial |
| 22:27.27 | WIMPy | But you should know that already. |
| 22:30.42 | dan_j | argh. sorry. i keep forgetting that its documented there |
| 22:40.05 | WIMPy | Slightly off topic, but essential fro VOIP: Does anyone know since when Linux tc is broken? |
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| 23:06.07 | ChannelZ | Broken how? |
| 23:06.29 | ChannelZ | It is complicated and mysterious for sure |
| 23:06.45 | phix | Linux tc? |
| 23:06.51 | ChannelZ | yes |
| 23:06.59 | phix | What's that again? |
| 23:07.05 | ChannelZ | Traffic control |
| 23:07.10 | phix | ah |
| 23:07.16 | phix | iproute2 stuff |
| 23:07.26 | ChannelZ | Yah |
| 23:07.38 | phix | Yeah, it is broken if you don't compile the kernel modules |
| 23:08.31 | WIMPy | It doesn't seem to do anything anymore. |
| 23:08.43 | phix | I actually haven't looked into tc for a few years now, is there a decent module for limiting bandwidth? or do I just discard every x packet and hope the TCP protocol kicks in and limits the bandwidth? |
| 23:09.07 | WIMPy | It used to do real wonders in the past. But now I cot complaints about bad voice quality while uploading. |
| 23:09.39 | ChannelZ | It works for me, at least in as much as I have it configured correctly |
| 23:09.45 | phix | tweak your settings? perhaps RTP is using a port you are not setting a higher priority to |
| 23:09.55 | WIMPy | So I decreased to maximum bandwidth but had to see that it sends much faster than I allowed. |
| 23:10.28 | phix | oh, you limit speed? or you specify a weighting / priority system on other packets? |
| 23:10.29 | WIMPy | So it doesn't seem to have any effect any more. |
| 23:10.29 | ChannelZ | What types of queues are you using? |
| 23:10.45 | WIMPy | You can't do without a limit. |
| 23:11.02 | phix | you can control upload with priority queues |
| 23:11.14 | ChannelZ | You limit speed, and sometimes drop packets to get people sending you crap to slow down |
| 23:11.19 | WIMPy | I use htb and sfq. |
| 23:11.43 | WIMPy | The important thing is to get things reordered. |
| 23:11.44 | phix | ChannelZ: so you still need to drop packets to control download? no TCP window manipulation modules yet? |
| 23:11.57 | WIMPy | The important stuff like rtp should cut in line when sending. |
| 23:12.04 | ChannelZ | There might be, but it's the easiest way |
| 23:12.27 | ChannelZ | WIMPy: Did the interface change? Maybe your rules are all on the wrong one? :) |
| 23:12.30 | phix | ChannelZ: easy but inefficient, especially when you are paying by the Gb |
| 23:12.40 | WIMPy | No, it didn't change. |
| 23:12.57 | WIMPy | And I just changed it manually while testing. |
| 23:13.10 | phix | (Which I am on edge / HTSPDPSDPSPSPPPPSSDD) |
| 23:13.50 | phix | oh noes! is Asterisk 11 at end of life? |
| 23:14.05 | phix | or is the topic the release dates? |
| 23:14.18 | ChannelZ | 1.4 is, but people still use it |
| 23:14.35 | phix | ChannelZ: I used 1.6 and 1.8 for ages. I am on 11 now |
| 23:15.02 | ChannelZ | Yeah I run 11 at work. 12 for fun at home |
| 23:15.30 | phix | I run 11 for work, home and clients :) |
| 23:15.47 | phix | so if it is going end of life I should probably consider updating it |
| 23:17.09 | ChannelZ | But no 11 isn't EOL |
| 23:17.32 | ChannelZ | Security fixes through 2016, EOL in 2017 |
| 23:17.35 | phix | phew |
| 23:17.53 | phix | ok great, I have 2.something years :) |
| 23:18.31 | phix | so that answers my previous question, the dates in the topic are release dates not eol dates |
| 23:18.34 | phix | :) |
| 23:18.39 | ChannelZ | Here's the chart: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
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| 23:21.21 | WIMPy | Strange. According to tc -s qdisc show I did have packets OverLim. |
| 23:23.33 | WIMPy | But none dropped. Is that the way it should look like? |
| 23:23.56 | WIMPy | can't remember. It's too long ago I looked into that. |
| 23:25.31 | WIMPy | Great if you find an old box to compare. Yes, it looks the same. So why doesn't it have anny effect any more? |
| 23:27.49 | ChannelZ | You can try something like watch -n1 tc -s class ls dev eth0 (or whatever interface) to see if traffic is going where you think it is. |
| 23:28.35 | ChannelZ | I wish there was a better way to evaluate it (maybe/probably there is and I just don't know it.) How are you selecting packets for priority? |
| 23:28.43 | ChannelZ | Maybe that's what broke |
| 23:29.27 | WIMPy | By iptables marks. But the queues seem to be all in use. So that should be fine. |
| 23:30.20 | WIMPy | Your line shows the same. |
| 23:30.38 | WIMPy | So it seems to be only the limit that doesn't work. |
| 23:31.37 | ChannelZ | Sorry I'm no expert. It's all new to me within the last few weeks, I've only just gotten something workable running. |
| 23:32.00 | WIMPy | With you command, the overlimits are 0, however. |
| 23:32.07 | WIMPy | your |
| 23:32.44 | ChannelZ | I only see overlimits on the root. No idea what it means |
| 23:33.17 | WIMPy | Yes, that's ok. |
| 23:33.50 | WIMPy | But I only see them with tc -s qdisc show, not with tc -s class ls. |
| 23:34.07 | ChannelZ | Right yes, same here (sorry that was unclear) |
| 23:34.17 | WIMPy | ok |
| 23:34.24 | WIMPy | Hmm |
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| 23:36.31 | ChannelZ | FWIW - http://pastebin.com/CWfxVVtr |
| 23:38.31 | WIMPy | Well, that's interesting. |
| 23:38.45 | WIMPy | rate and pps stay 0 for me. |
| 23:39.50 | ChannelZ | With activity happening? Those should be 'current' rates |
| 23:40.21 | ChannelZ | like 1:10 is my voip/ack high priority class, and nothing much is going on at the moment |
| 23:40.29 | WIMPy | The the sent bytes and pkt count happily, but the rates stay 0. |
| 23:40.41 | ChannelZ | strange |
| 23:40.45 | WIMPy | So that must be the problem. |
| 23:41.03 | WIMPy | Now the interesting question is what's going on and/or why... |
| 23:41.14 | ChannelZ | Is your script crazy or pasteable? |
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| 23:44.45 | WIMPy | http://wimpy.yeti.dk/pastebin |
| 23:45.12 | WIMPy | That's without the iptables stuff. But that's not the point here. |
| 23:45.41 | WIMPy | It's was based upon an example from the LARTC. |
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| 23:50.47 | ChannelZ | Hmmm. I don't really see anything much different from mine |
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| 23:51.26 | WIMPy | Maybe something hase gone missing from the kernel. I'm currently checking the config there... |
| 23:53.15 | ChannelZ | Mine: http://pastebin.com/JgkhRbZj |
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