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| 00:58.19 | drynish | I really like asterisk |
| 00:58.26 | drynish | I had to change my echo canceller |
| 00:58.40 | drynish | I was using Oslec and I find out that mg2 was doing a better job |
| 00:58.45 | drynish | is it normal? |
| 01:04.22 | Kobaz | some are better than others |
| 01:09.51 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 01:10.15 | drynish | yeah but every review I read, it seems oslec the best ovreall |
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| 01:18.42 | leifmadsen | it really depends on the situation -- you use what works in your configuration |
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| 01:25.50 | jackmcbarn | jeffspeff: http://pastebin.com/QLnK1wGX |
| 01:25.53 | *** mode/#asterisk [+b help!*@*] by leifmadsen |
| 01:26.02 | *** mode/#asterisk [-b help!*@*] by leifmadsen |
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| 02:56.34 | pabelanger | ~newbook |
| 02:56.34 | infobot | Please see ~thebook for more information about Asterisk: The Definitive Guide |
| 02:56.39 | pabelanger | ~thebook |
| 02:56.39 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 02:58.02 | djuhl30 | http://paste.pocoo.org/show/453481/ |
| 02:58.14 | djuhl30 | http://paste.pocoo.org/show/453488/ |
| 02:58.24 | djuhl30 | http://paste.pocoo.org/show/453490/ |
| 02:58.57 | djuhl30 | http://paste.pocoo.org/show/453491/ |
| 02:59.07 | djuhl30 | http://paste.pocoo.org/show/453492/ |
| 02:59.15 | djuhl30 | http://paste.pocoo.org/show/453493/ |
| 02:59.26 | djuhl30 | http://paste.pocoo.org/show/453494/ |
| 02:59.35 | djuhl30 | http://paste.pocoo.org/show/453495/ |
| 02:59.47 | djuhl30 | I was able to receive a call once... |
| 02:59.57 | djuhl30 | I can't receive calls now. |
| 03:00.06 | djuhl30 | And I don't know how to read all this. |
| 03:00.11 | djuhl30 | Can anyone help? |
| 03:00.26 | djuhl30 | I turned debugging on. |
| 03:01.25 | *** join/#asterisk blixen (~blixen@ppp121-45-198-220.lns20.cbr1.internode.on.net) |
| 03:01.38 | djuhl30 | I had to paste in seperate files.. I used wgetpaste |
| 03:03.21 | djuhl30 | SIP/2.0 404 Not Found I can't tell what isn't found |
| 03:06.59 | djuhl30 | <PROTECTED> |
| 03:07.20 | djuhl30 | I don't know if that is a problem or not |
| 03:13.57 | ChannelZ | I don't know what you have going on |
| 03:15.51 | ChannelZ | I don't see any INVITEs. The 'not found' is in response to the OPTIONS though I dont know why |
| 03:30.40 | djuhl30 | cool |
| 03:37.44 | ChannelZ | if I had to guess your peer entries/registration in sip.conf is totally not right (but that's a fairly easy guess with no particulars) |
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| 03:42.39 | raden | Katty, ???? |
| 03:43.52 | ChannelZ | unlikely |
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| 07:03.56 | gdarcan | anybody help |
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| 07:12.40 | gdarcan | I have a problem with DigiumTDM800P, maybe a firmware or driver problem or just only a configuration problem. Is it possible to solve it here with guidance of someone ? Or am I totally at wrong place? Please somebody answer yes or no... |
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| 07:22.23 | ChannelZ | arf |
| 07:23.49 | ChannelZ | gdarcan: what seems to be the problem |
| 07:24.37 | gdarcan | hi |
| 07:24.52 | gdarcan | problem is : sip kernel: wctdm24xxp 0000:11:00.0: Host failed to service card interrupt within 128 ms which is a hardunderun. |
| 07:25.16 | gdarcan | system 50% works, 50% doesn't work |
| 07:25.47 | gdarcan | sometimes when somebody tries to call outside, it says "all the channels are busy" |
| 07:26.13 | gdarcan | is this a hardware issue ? |
| 07:31.16 | ChannelZ | probably |
| 07:31.25 | ChannelZ | Can you move it to a different PCI slot? |
| 07:31.40 | gdarcan | yes, I can try |
| 07:32.06 | gdarcan | after move, are there any conf files should I change ? |
| 07:32.08 | ChannelZ | I wonder if the slot it's in is sharing with something else and causing lousy timing |
| 07:32.15 | ChannelZ | Shouldn't be, no |
| 07:32.45 | gdarcan | do you want any command outputs ? |
| 07:33.29 | gdarcan | lspci , lsdahdi etc. |
| 07:34.09 | ChannelZ | you might take note of lspci but it's not necessarily important yet |
| 07:34.35 | gdarcan | ok, I'm trying now if you'll be here within 5 minutes.. |
| 07:35.17 | ChannelZ | yah |
| 07:35.35 | ChannelZ | actually |
| 07:35.41 | ChannelZ | a couple of questions |
| 07:35.52 | ChannelZ | Do you have the echo cancellation card? |
| 07:36.26 | gdarcan | actually I don't know if there'is a card |
| 07:36.33 | gdarcan | how can I know ? |
| 07:36.45 | gdarcan | 11:00.0 Ethernet controller: Digium, Inc. Wildcard TDM800P 8-port analog card (rev 11) |
| 07:36.45 | gdarcan | <PROTECTED> |
| 07:36.45 | gdarcan | <PROTECTED> |
| 07:36.45 | gdarcan | <PROTECTED> |
| 07:36.45 | gdarcan | <PROTECTED> |
| 07:37.02 | gdarcan | [root@sip ~]# lsdahdi |
| 07:37.03 | gdarcan | ### Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER) |
| 07:37.03 | gdarcan | <PROTECTED> |
| 07:37.03 | gdarcan | <PROTECTED> |
| 07:37.03 | gdarcan | <PROTECTED> |
| 07:37.08 | ChannelZ | please use pastebin |
| 07:37.09 | ChannelZ | ~pb |
| 07:37.10 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 07:37.14 | gdarcan | ok sorry |
| 07:37.57 | ChannelZ | for fun pastebin your /proc/interrupts in its current config |
| 07:40.12 | gdarcan | http://pastebin.com/uVFWY8C7 |
| 07:41.18 | ChannelZ | hmm. |
| 07:42.09 | ChannelZ | stop Asterisk and run dahdi_test |
| 07:43.16 | ChannelZ | What percentages do you get? |
| 07:43.35 | gdarcan | it's running, should I CTRL+C it somehere or will it stop ? |
| 07:43.51 | gdarcan | 100.000% 99.995% 99.999% 99.999% 99.999% 99.830% 99.827% 99.999% 99.999% 99.998% 100.000% 99.998% |
| 07:44.25 | ChannelZ | yeah it'll just go forever |
| 07:44.30 | gdarcan | Best: 100.000 -- Worst: 88.065 -- Average: 99.800526 |
| 07:44.51 | gdarcan | there's something in inconsistent state I think |
| 07:45.34 | ChannelZ | well that you hit 100% is interesting. I never get better than 99.998% usually |
| 07:45.54 | ChannelZ | The 88% is a little odd |
| 07:46.39 | ChannelZ | What version of DAHDI? |
| 07:48.05 | gdarcan | how can I know ? |
| 07:49.33 | gdarcan | what command should run ? |
| 07:51.00 | ChannelZ | sorry wandered off |
| 07:51.14 | ChannelZ | try grep dahdi /var/log/messages |
| 07:51.45 | gdarcan | 2.4.1.2 |
| 07:52.01 | ChannelZ | hmm |
| 07:52.44 | ChannelZ | Well at the end of all this it seems to just not be responding to interrupts consistently, why that is is hard to say. Easiest thing to try though is move the card into a different slot and see if it behaves any better. |
| 07:53.12 | gdarcan | ok I'll try it asap. |
| 07:53.33 | gdarcan | after the staff finishes talking with customers... |
| 07:54.11 | ChannelZ | so it's in service and functional, you're just getting those warnings once in awhile? |
| 07:55.48 | gdarcan | tail /var/log/messages is at http://pastebin.com/5h70AZ47 |
| 07:56.42 | ChannelZ | ok so its not so occasional |
| 07:57.32 | gdarcan | but when the outbound channels stuck it leaves as it is till I restart everything incl. FCTs |
| 07:57.55 | gdarcan | may an FCT cause this ? |
| 07:58.58 | ChannelZ | FCT? |
| 07:59.51 | gdarcan | we insert a GSM simcard, there's an antenna on it and a standart RJ11 phoneline otherside connecting to TDM800P where outbound calls go through it |
| 08:00.48 | gdarcan | I find out meaing at google, Fixed Cellular Terminal |
| 08:01.20 | gdarcan | GMS Gateway in English wikipedia |
| 08:01.48 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 08:01.55 | gdarcan | we name those devices FCT in Turkey... |
| 08:02.06 | ChannelZ | I doubt it has anything to do with it |
| 08:02.51 | gdarcan | may a socket connection to port 5038 cause all ? |
| 08:03.16 | gdarcan | a php script writes the hangup dial DTMF activities to database realtime |
| 08:03.21 | ChannelZ | no that's the Manager interface |
| 08:04.20 | *** join/#asterisk gravin (~gravin@113.210.253.234) |
| 08:04.36 | ChannelZ | The communication between the card and the hardware is not consistent and fast enough sometimes. Unless you've got other tasks running on the machine causing the hardware interrupts to be ignored... |
| 08:05.12 | gdarcan | hmm |
| 08:06.17 | ChannelZ | Is this a new setup or has the system been running and this problem just started recently? |
| 08:06.17 | gdarcan | that php script may cause this then |
| 08:06.43 | gdarcan | this is a new systems we just switched to this last week |
| 08:07.21 | ChannelZ | It's doubtful a PHP script could be consuming so many resources it's screwing up the card. |
| 08:07.49 | gdarcan | I'll move that script outside the machine... |
| 08:08.06 | ChannelZ | unless you're doing something unusual or it's badly written and sitting in a busy loop screwing up the whole system |
| 08:08.20 | gdarcan | by the way it is very handy, we got popup on new call and see the DTMFs |
| 08:08.49 | ChannelZ | I mean, your dahdi_test results tell most of the story |
| 08:08.53 | gdarcan | yes it was screwing up the system I think |
| 08:10.55 | ChannelZ | seems odd. Is it sending thousands of commands a second or something? |
| 08:11.30 | gdarcan | it runs while (1==1) |
| 08:11.47 | ChannelZ | doing what? |
| 08:11.51 | gdarcan | infinite loop listening socket messages line by line |
| 08:12.32 | gdarcan | when catches Event: Dial or Event: Hangup, Event: Bridge etc. puts them on database |
| 08:13.05 | gdarcan | logged in personnel sees a popup when somebody calls him |
| 08:13.53 | ChannelZ | are you using non-blocking IO or something? |
| 08:14.20 | gdarcan | what is non-blocking IO ? |
| 08:15.01 | gdarcan | my boss wanted a system where we give a PIN code to customers when we need security confirmation we say them dial PIN code |
| 08:15.13 | ChannelZ | Assuming your PHP script is using sockets to talk to Manager, it'd be like set_socket_blocking(0) |
| 08:16.12 | ChannelZ | or stream_set_blocking now I guess |
| 08:16.44 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 08:16.55 | ChannelZ | If your PHP script is indeed causing your problems, it's very poorly written and needs to be looked at |
| 08:17.18 | gdarcan | yes it is written in 1 day after I discoverd manager interface |
| 08:17.41 | ChannelZ | if you're using non-blocking IO to read and sitting in a loop calling a read function which is returning immediately when there's no data, you're consuming tons of CPU time doing nothing |
| 08:17.45 | gdarcan | there was a limited time to finish this |
| 08:17.59 | gdarcan | yeahhh |
| 08:18.39 | gdarcan | I'm pastebinning it |
| 08:22.45 | gdarcan | f..ing php script is at http://pastebin.com/337Cajf0 |
| 08:23.45 | gdarcan | smoking break while you are looking and swearing me, i'll comeback in 4mins ;) |
| 08:24.24 | gdarcan | by the way I cut some different event catching parts in the loop |
| 08:24.31 | gdarcan | dial, hangup, bridge |
| 08:25.01 | gdarcan | dtmf and unlink |
| 08:25.33 | ChannelZ | well it's also cut off from long lines from pico or nano or whatever, but I get the idea |
| 08:32.51 | ChannelZ | I think blocking is turned on by default in PHP, at least it is for me.. |
| 08:33.18 | ChannelZ | In which case I don't think this script can be causing your problem being in a true busy-loop |
| 08:34.49 | ChannelZ | Is your load average high while it's running? |
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| 08:35.51 | ChannelZ | (my turn to leave, going on a taco run) |
| 08:39.39 | gdarcan | no |
| 08:39.46 | gdarcan | there's a bug maybe... |
| 08:40.05 | gdarcan | it runs with 0 cpu load for an hour |
| 08:40.14 | gdarcan | then it jumps to 100% |
| 08:45.12 | gdarcan | full version of the script is: http://pastebin.com/fFiazCtP |
| 08:47.18 | gdarcan | Even though I stopped the script running, I got the same error again: Aug 6 11:21:54 sip kernel: wctdm24xxp 0000:11:00.0: ERROR: Unable to service card within 25 ms and unable to further increase latency. |
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| 09:05.21 | ChannelZ | hmm. |
| 09:09.41 | *** join/#asterisk gdarcan (~gdarcan@support.vital.net.tr) |
| 09:12.13 | gdarcan | hi ChannelZ i've disconnected and reconnected did you write something after "hmmm" |
| 09:12.39 | ChannelZ | nope |
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| 10:03.10 | [sr] | howdy |
| 10:03.26 | [sr] | how/where do i disable the timeout from call's onhold |
| 10:03.28 | [sr] | of 5 minutes? |
| 10:12.51 | *** join/#asterisk cyborg-one (1000@188-115-162-27.broadband.tenet.odessa.ua) |
| 10:14.25 | ChannelZ | hmm I don't think there is one |
| 10:14.40 | ChannelZ | That might be on your device |
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| 10:26.13 | [sr] | ChannelZ: on the fone? |
| 10:27.52 | [sr] | or the sip extension.. let me see |
| 10:28.14 | ChannelZ | the actual phone |
| 10:28.14 | *** part/#asterisk esperanto (~rusty@pool-96-224-237-56.nycmny.fios.verizon.net) |
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| 10:28.20 | *** part/#asterisk esperanto (~rusty@pool-96-224-237-56.nycmny.fios.verizon.net) |
| 10:28.42 | ChannelZ | I don't believe there is any timeout in Asterisk for putting people on hold. There is for parking lots |
| 10:29.16 | [sr] | ChannelZ: the thing is, when i use onhold, or on a transfer, the call terminates in 5 minutes |
| 10:29.27 | [sr] | it's an exact value |
| 10:29.38 | [sr] | but i'm going to check both sip extension and phone settings |
| 10:29.44 | [sr] | i'll let you know |
| 10:29.46 | ChannelZ | With verbose on, what does the console say? |
| 10:30.11 | [sr] | let me do that test 1st then |
| 10:30.16 | [sr] | just wait 5m :) |
| 10:30.42 | *** join/#asterisk djuhl30 (~quassel@121.135.82.142) |
| 10:31.12 | djuhl30 | ChannelZ: how? |
| 10:31.39 | ChannelZ | how what? |
| 10:32.18 | djuhl30 | I'll paste my sip.conf and extensions.conf |
| 10:32.30 | ChannelZ | ...ok... why? |
| 10:36.19 | djuhl30 | Because you said "if I had to guess your peer entries/registration in sip.conf is totally not right (but that's a fairly easy guess with no particulars)" |
| 10:36.57 | djuhl30 | I'll have to change the password or blot it out |
| 10:37.25 | ChannelZ | oh. That was like 6 hours ago |
| 10:37.35 | djuhl30 | I had to go back to work |
| 10:37.39 | djuhl30 | I have a day job |
| 10:38.09 | djuhl30 | I am stationed in South Korea |
| 10:38.15 | ChannelZ | I get it - you just picked up a conversation like time had stopped, I didn't know what you were even talking about |
| 10:38.26 | ChannelZ | no matter |
| 10:38.49 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 10:38.55 | [sr] | ops back |
| 10:39.03 | djuhl30 | You want to take a look? |
| 10:40.25 | [sr] | ChannelZ: what are we looking for? i dont see nothing that has the tmieout value |
| 10:41.07 | ChannelZ | [sr]: just pastebin the whole call |
| 10:41.11 | ChannelZ | djuhl30: sure |
| 10:41.14 | djuhl30 | http://paste.pocoo.org/show/453596/ |
| 10:41.51 | [sr] | ChannelZ: http://pastebin.com/Fkyr89px |
| 10:42.07 | ChannelZ | djuhl30: and what was the problem/error you were actually having? I don't remember |
| 10:43.18 | djuhl30 | I can't do a test call to my did I bought with sip route sip 2.0 Not found |
| 10:43.51 | [sr] | ChannelZ: the last lines show's something interesting... on the sip debug |
| 10:43.58 | [sr] | last 5 lines |
| 10:45.04 | [sr] | thats when the call hangs up, |
| 10:45.39 | [sr] | let me see if it hangs with the same message |
| 10:47.45 | ChannelZ | there's a lot of noise in there (and that's not a SIP debug) but it seems like SIP/20 is hanging up. |
| 10:48.35 | [sr] | this is a lan connection |
| 10:48.53 | [sr] | in the end it says there's no CID, which is true |
| 10:49.24 | ChannelZ | djuhl30: What are you calling from? And can you pastebin the console output from the call attempt or are you not even getting anything on Asterisk |
| 10:50.51 | [sr] | ChannelZ: ok, hangup with the same thing, with the message "that's odd..... got response on a call we didn't know about" |
| 10:50.55 | ChannelZ | [sr]: Oh, here's your problem |
| 10:50.58 | ChannelZ | NOTICE[2446] chan_sip.c: Disconnecting call 'SIP/69-00000092' for lack of RTP activity in 301 seconds |
| 10:51.08 | [sr] | and the other that sayd there's no CID |
| 10:51.09 | ChannelZ | Do you actually have music set? |
| 10:51.24 | [sr] | ChannelZ: yes |
| 10:52.19 | djuhl30 | ChannelZ: Give me a sec. I don't know how to read the debug output |
| 10:52.30 | ChannelZ | [sr]: look at "rtpholdtimeout" in sip.conf |
| 10:53.04 | ChannelZ | djuhl30: don't turn on debug |
| 10:53.16 | [sr] | ChannelZ: bingo :$ |
| 10:53.18 | ChannelZ | djuhl30: just do 'core set verbose 3' |
| 10:53.23 | djuhl30 | ok |
| 10:54.02 | [sr] | ChannelZ: ok thanks, found the mistery! |
| 10:54.22 | djuhl30 | <PROTECTED> |
| 10:54.22 | djuhl30 | <PROTECTED> |
| 10:54.22 | djuhl30 | <PROTECTED> |
| 10:54.29 | djuhl30 | 3 lines |
| 10:54.39 | djuhl30 | 4 |
| 10:54.58 | ChannelZ | hmm that's odd |
| 10:55.56 | djuhl30 | me? |
| 10:56.02 | ChannelZ | yeah |
| 10:56.24 | djuhl30 | How do I troubleshoot this? |
| 10:56.26 | ChannelZ | Your dialplan runs Answer but then immediately dies |
| 10:56.41 | ChannelZ | pastebin a sip debug of the call attempt |
| 10:56.46 | ChannelZ | sip set debug on |
| 10:56.49 | ChannelZ | make the call |
| 10:56.50 | ChannelZ | sip set debug off |
| 10:56.57 | djuhl30 | I got it |
| 10:58.16 | djuhl30 | http://pastebin.com/FmRYm1Eu |
| 11:01.23 | ChannelZ | I'm a little confused |
| 11:01.48 | ChannelZ | You're trying to call in to your flowroute DID from where? |
| 11:02.02 | ChannelZ | I see an Android device on your LAN |
| 11:10.55 | ChannelZ | I think partially your NAT settings are wrong.. from what I can tell your Asterisk is giving Flowroute a LAN IP address to connect to. Which they probably ignore, but then they send a new invite to a totally different IP and then things get weird from there |
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| 11:18.30 | ChannelZ | guess you got busy on other things. I gotta go to bed |
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| 12:00.27 | djuhl30 | Sorry I had to give attention to the wife |
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| 12:12.51 | djuhl30 | I'll never figure this out |
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| 13:27.18 | jackmcbarn | im having lots of weird issues with my dialplan. it's at http://pastebin.com/QLnK1wGX and i listed the issues as comments |
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| 13:50.28 | leifmadsen | jackmcbarn: your 'i' extension doesn't go back to where DTMF is listened for, so it hangs up |
| 13:50.56 | leifmadsen | and after the timeout period if nothing has happened, and there is nothing being programmed to tell the call where to go, yes, it will hang up |
| 13:51.29 | leifmadsen | there is no difference between those pbx-invalid and invalid sounds (you can look in the .txt file included with the sounds package to see what they should say |
| 13:51.30 | leifmadsen | ) |
| 13:51.56 | leifmadsen | Additionally, no DTMF would ever be listened for because you're only using Playback(), which does not listen for DTMF -- you need to use Background() |
| 13:52.30 | leifmadsen | also, your authentication seems like it must be incorrect (unless you're trying to accept unauthenticated calls) because you should never use [default] as your primary context. |
| 13:52.32 | leifmadsen | ~thebook |
| 13:52.32 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 13:52.49 | leifmadsen | Many of these questions can be answered in the first few chapters of that book. |
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| 14:08.26 | Kobaz | i do a context default { Hangup(); } so that I don't get 404 sip errors |
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| 14:19.50 | leifmadsen | aye |
| 14:19.57 | leifmadsen | looks like AEL :) |
| 14:28.26 | Kobaz | it does, doesn't it |
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| 14:32.02 | Kobaz | i just wrote a DumpChans app |
| 14:32.12 | Kobaz | it's really handy |
| 14:33.20 | Kobaz | give me another year or so, and you'll get access to all the asterisk internals through dialplan |
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| 16:24.06 | jackmcbarn | leifmadsen: the i extension works like i want it to. its the 1001 extension that i want to listen for dtmf. why isnt it working? isnt WaitExten supposed to listen for DTMF? im not entering the numbers until the playback is finished |
| 16:24.28 | jackmcbarn | right now, im sending all calls into the test dialplan. once i get these issues solved, i'll set up proper authentication and contexts |
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| 16:33.27 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 16:33.29 | [sr] | hi WIMPy |
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| 16:47.55 | [sr] | damn i'm sleepy |
| 16:56.12 | [sr] | brb |
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| 18:01.36 | BugKhaM | I am calling from "A" party iax2 friend to another "B" party iax2 friend in a different server, on iax2 debugging mode I see the "A" party callerid in the "CALLING NUMBER" portion |
| 18:02.06 | BugKhaM | in which channel variable is it stored? |
| 18:03.17 | BugKhaM | ${CALLERID(ani2)} ? |
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| 18:37.55 | raden | Katty, ~!!~ |
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| 19:10.23 | Cadey | why is asterisk going to 10. and not 2.0 out of interest? :) |
| 19:11.06 | WIMPy | 10=2 |
| 19:11.15 | Cadey | oh lol |
| 19:11.23 | Cadey | and here was me thinking in decimal |
| 19:11.57 | WIMPy | Well, you know that one about 10 kind of people, don't you? |
| 19:12.13 | Cadey | lol who has'nt :P |
| 19:12.18 | WIMPy | But I don't think the subversions will be binary as well. |
| 19:12.38 | Cadey | I should have been thinking in geek terms :D |
| 19:13.26 | WIMPy | Version 10.10110.101 |
| 19:14.17 | Cadey | 2.22.5 ? |
| 19:15.01 | WIMPy | That's what I said :-) |
| 19:15.10 | Cadey | :P |
| 19:15.30 | Cadey | speaking of binary ive spent the last 3 days redoing out network |
| 19:15.48 | Cadey | 3 days of cisco configs, arg |
| 19:15.48 | WIMPy | You mean 11 days? |
| 19:15.52 | Cadey | lol |
| 19:16.34 | Cadey | real pain, multihomed network with 8 networks onit |
| 19:16.44 | Cadey | 8 internal |
| 19:17.01 | WIMPy | Ah, multihomed can be fun. |
| 19:17.12 | Cadey | or not :) |
| 19:17.25 | Cadey | I do all the routing on the router amazingly enough :P |
| 19:17.34 | WIMPy | Asterisk doesn't like it, either. |
| 19:17.46 | Cadey | well asterisk doesnt no its happening |
| 19:18.14 | Cadey | as its all done at the router, I use SLA's to manage route redundency |
| 19:18.17 | WIMPy | Then hope it doesnt find out. |
| 19:18.39 | Cadey | SLA+Track==Win |
| 19:19.09 | Cadey | should take his CCNP |
| 19:22.16 | Cadey | has anyone done much with vid on asterisk? |
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| 19:22.38 | Cadey | ive never really looked into it tbh but softphones all support it so could be fun to get working :P |
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| 19:29.00 | Cadey | seems its pick your codec and wack videosupport=yes to the sip config of the extensions and away you go :) |
| 19:29.01 | Cadey | simples |
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| 19:37.29 | ChannelZ | yawns |
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| 20:40.54 | [sr] | one question |
| 20:41.04 | [sr] | the 10.0.0 version shoundn't be 1.10 ? |
| 20:41.25 | p3nguin | Apparently not. |
| 20:41.36 | p3nguin | That was easy. NEXT! |
| 20:41.49 | [sr] | oki :) |
| 20:42.04 | [sr] | it's 'cause i saw on some places a few days ago that the version would be 1.10 |
| 20:42.13 | p3nguin | They were wrong. |
| 20:42.18 | [sr] | but yes, asterisk is mature enought to be 10.x |
| 20:42.48 | Cadey | lol |
| 20:42.58 | Cadey | we had this convo about an hour ago |
| 20:43.03 | Cadey | we came to the conclusion |
| 20:43.15 | Cadey | 1.0, 1.2, 1.4, 1.6, 1.8, 10 |
| 20:43.15 | p3nguin | It'll just keep happening. |
| 20:43.18 | Cadey | 10 being 2 |
| 20:43.34 | p3nguin | 10 being 1.10 |
| 20:43.37 | p3nguin | not 2 |
| 20:44.13 | Cadey | no, 10 being 2 |
| 20:44.23 | Cadey | like 110 is 6 |
| 20:44.35 | p3nguin | So we're talking in binary, now? |
| 20:44.39 | Cadey | yeah |
| 20:44.48 | p3nguin | Wonderful! |
| 20:45.06 | Cadey | me and WIMPy had the convo |
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| 20:59.38 | Bipul | p3nguin, i won the Presentation competation |
| 20:59.48 | Bipul | and i am thinking to joine asterisk workshop |
| 20:59.49 | Bipul | also |
| 21:00.06 | Bipul | Thanks for you great help ! |
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| 21:41.55 | [sr] | going to sleep |
| 21:43.18 | ChannelZ | nighty |
| 21:44.05 | [sr] | merci |
| 21:44.08 | [sr] | be well |
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| 23:14.14 | gruvfunk | greets all |
| 23:17.58 | gruvfunk | having an interesting situation here and wondering if anyone could shed some light or point in the right direction |
| 23:18.37 | gruvfunk | a customer just called complaining that his ITSP bills have suddenly increased, checking logs there are a great number of unauthorized calls |
| 23:19.22 | gruvfunk | Asterisk 1.8 with a dozen or so service representatives making and taking calls, each registered to the PBX via Extension and complex password |
| 23:19.59 | gruvfunk | being that it's Saturday, their office is closed, and there is absolutely no one registered to the PBX |
| 23:20.18 | gruvfunk | yet, I'm watching as numerous calls come in via a strange IP address |
| 23:21.09 | gruvfunk | beyond traditional security measures (patching, resetting passwords, sanitizing sip.conf), can anyone shed some light on this? |
| 23:21.37 | gruvfunk | How are calls being placed to this PBX without User Registration? |
| 23:24.22 | ChannelZ | Do you have 'allowguest=yes' on in sip.conf? |
| 23:24.52 | ChannelZ | if so they are 'unauthenticated' calls |
| 23:25.32 | ChannelZ | and/or your dialplan is such that they're making outgoing calls |
| 23:25.58 | gruvfunk | checking |
| 23:26.06 | ChannelZ | pastebin some of your console output if you want |
| 23:26.25 | gruvfunk | yep, ;allowguest=no (default is yes) |
| 23:26.27 | p3nguin | Yay for allowing inbound calls to dial outbound! |
| 23:26.44 | ChannelZ | ok. So then it's probably just a bad dialplan and someone figured it out |
| 23:26.45 | gruvfunk | yeah, rough... closing the hole |
| 23:27.18 | ChannelZ | I assume they're coming in on SIP and dialing back out SIP through your system |
| 23:27.19 | p3nguin | Not only did you allow anonymous calls in, but you're using the default context for outgoing calls. Bad! |
| 23:27.33 | gruvfunk | not my config |
| 23:27.38 | gruvfunk | i'm just helping fight fires |
| 23:27.46 | ChannelZ | uncomment allowguest=no for starters |
| 23:27.49 | gruvfunk | yep |
| 23:28.15 | gruvfunk | context wise, it is not using default for outgoing, that's all looks legit and carved out |
| 23:28.27 | gruvfunk | i'm guessing ChannelZ's first guess is spot on |
| 23:29.16 | ChannelZ | You should be able to tell how they're coming in; It sort of doesn't matter if it's due to an anonymous call or they are dialing in your regular number(s), if the dialplan is allowing them to dial back out. |
| 23:30.58 | p3nguin | Yeah, the problem isn't allowing anonymous calls. I allow anonymous calls. |
| 23:31.10 | ChannelZ | yah |
| 23:31.14 | p3nguin | But inbound calls of any sort do not have access to go back out. |
| 23:31.37 | p3nguin | At least not without knowing the secret access stuff. |
| 23:32.00 | ChannelZ | gruvfunk: does extensions.conf use DISA() anywhere? |
| 23:32.32 | p3nguin | I'd guess there's an include where it doesn't belong. |
| 23:32.52 | ChannelZ | yeah, or a Goto someplace |
| 23:33.58 | gruvfunk | they were definitely not dialing one of our numbers |
| 23:34.14 | gruvfunk | they were coming in as anonymous calls from a specific IP address... |
| 23:34.30 | p3nguin | But that alone does not cost you money. |
| 23:34.31 | ChannelZ | no matter |
| 23:34.35 | gruvfunk | no DISA |
| 23:34.47 | ChannelZ | So your dialplan is just a little bad then |
| 23:34.57 | gruvfunk | oh I see what you mean |
| 23:35.08 | gruvfunk | they guessed on the out context |
| 23:35.13 | gruvfunk | and therefore it must be set to "default" |
| 23:35.16 | ChannelZ | You should be able to figure it out by watching the console with verbose on and seeing where things are going, but pastebin your extensions.conf if you like |
| 23:35.19 | gruvfunk | back to digging through this |
| 23:35.43 | ChannelZ | I actually (maybe) need to re-think my own dialplan a little. My [incoming] has extens for a bunch of phones, but to dial them I do a Goto into my [internal] context which does the Dial()ing and handles voicemail, etc. But there's a bunch of other extensions in there. |
| 23:35.47 | p3nguin | You could also pastebin your entire dial plan for us to check for you. |
| 23:36.22 | ChannelZ | However, if I'm never doing a Background() or WaitExten or anything else that would cause * to be "listening" for an extension, no one should be able to access any of them I don't think |
| 23:37.07 | p3nguin | As long as you aren't doing pattern matches with the Goto()s, the odds are in your favor. |
| 23:37.24 | ChannelZ | no, they're all static. |
| 23:37.59 | p3nguin | It's hard to get to another extension when it's hard-coded as what you want it to be. |
| 23:38.14 | p3nguin | Unless, like you said, you allow other extens to be entered. |
| 23:38.17 | ChannelZ | I basically just didn't want to have to duplicate the Dials and status checking and voicemail stuff in both places |
| 23:38.48 | p3nguin | I do the same thing with Goto. I never considered it to be a problem. |
| 23:39.20 | ChannelZ | That said there's nothing much in my 'internal' context anyone would have any interest in even if they were able to do something by my own accident |
| 23:39.34 | ChannelZ | They could annoy a couple of specific people but that's about it :) |
| 23:39.37 | p3nguin | Any exten that goes somewhere else doesn't have the ability to dial out again. |
| 23:43.47 | Tim_Toady | thers also the posibility of code injection in the dialplan |
| 23:44.24 | ChannelZ | Yah, with patterns or variables getting used that somehow come from the caller |
| 23:44.28 | *** join/#asterisk fofware (~fabian@host106.186-109-251.telecom.net.ar) |
| 23:46.15 | Tim_Toady | 'README-SERIOUSLY.bestpractices.txt' doc in the asterisk source tree is always a good start :P |
| 23:47.04 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v000-202.mobile.uci.edu) |
| 23:48.32 | ChannelZ | indeedy |
| 23:56.43 | ChannelZ | Anyone have any good simple tutorials on LDAP? The stuff I've found in the past either goes off into the weeds or just has you do a bunch of stuff but doesn't explain what anything actually means. |