| 00:08.58 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 00:37.58 | saxa | got it , probably is this, http://www.grandstream.com/faqstroubleshooting.html#6 |
| 00:38.10 | saxa | now I need to see how to set up that |
| 00:38.17 | saxa | but thats for tomorrow |
| 00:38.19 | saxa | night |
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| 02:05.02 | Bloudermilk | Evening everyone. I was wondering if anybody had some advice for dynamically changing MoH classes while asterisk is running |
| 02:05.39 | Bloudermilk | I'm using MoH to play a music stream and I'd like to be able to dynamically change the stream to be played depending on the person calling |
| 02:06.00 | Bloudermilk | If it makes any difference, I am using AGI and AMI so I can control asterisk through there |
| 02:06.32 | Bloudermilk | Alternatively, if there's another method to play wav/ogg streams (from icecast) that would let me change the source dynamically that could work too |
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| 03:59.56 | niekie | Hehe. Boredom does things to people: http://afterthetone.org/ |
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| 07:41.10 | Elijah_ | Hi guys, got a cisco 7940 that won't seem to register. Phone's behind a NAT, asterisk has a public IP. Sip debug: http://pastebin.com/nVxiK51Y |
| 07:41.48 | Elijah_ | It seems to me asterisk just isn't replying back to the registers |
| 07:41.59 | Elijah_ | (Does work with xlite though, fwiw) |
| 07:43.40 | wdoekes2 | core set debug 1 |
| 07:43.54 | Elijah_ | It's set to 10, that's what I see |
| 07:44.07 | wdoekes2 | if it's at 10, you should see a lot more |
| 07:44.20 | wdoekes2 | probably 'debug' isn't directed to where you're looking |
| 07:44.23 | wdoekes2 | see logger.conf |
| 07:45.11 | Elijah_ | I'm looking in the asterisk console, where else would it go? |
| 07:45.26 | Elijah_ | console => notice,warning,error |
| 07:45.30 | Elijah_ | maybe add debug to that? |
| 07:46.05 | wdoekes2 | if you want it in your console, then yes |
| 07:46.19 | Elijah_ | ok... wow, yes a lot more stuff now |
| 07:46.22 | wdoekes2 | perhaps you want to set debug to a little less than 10 at first |
| 07:46.48 | Elijah_ | waiting for another register packet |
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| 07:47.43 | Elijah_ | Aha, here we go: http://pastebin.com/UeXQkYpZ |
| 07:49.20 | Elijah_ | But it looks like there's a From: tag to me... "From: sip:prescott@64.6.54.11" |
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| 08:02.21 | wdoekes2 | ergo, the cisco is broken |
| 08:02.36 | wdoekes2 | oh.. I read "no From: tag" |
| 08:02.42 | wdoekes2 | which would be correct |
| 08:02.57 | wdoekes2 | From: sip:prescott@64.6.54.11;tag=abcdef <-- from-tag |
| 08:03.46 | Elijah_ | aah yeah... didn't realize that had to be there |
| 08:04.44 | Elijah_ | Wonder how you would ever get the phone to add that |
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| 08:12.47 | wdoekes2 | firmware upgrade |
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| 08:18.29 | Elijah_ | Yeah, may try that. Thanks for the help though, at least we know what the trouble is |
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| 10:09.07 | pif | hi, any good or bad experience in using a users.conf vs sip/voicemail/iax.conf ? |
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| 11:18.49 | _zoom_ | hello |
| 11:19.01 | fenrus | yello |
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| 11:22.45 | _zoom_ | am trying to find a solution to monitor a set of asterisk peers, status, last call duration, total mins since last registrations using python |
| 11:22.57 | _zoom_ | is there anyone have already done this |
| 11:23.52 | fenrus | there is some nagios plugins for checking stuff like this |
| 11:24.04 | fenrus | might not be pyhton.. |
| 11:27.29 | _zoom_ | the reason for python cause am familiar with, but its ok any other powerful scripting language. |
| 11:29.26 | fenrus | i think ill just create some perl/bash hack to interface with my nagios that use nrpe |
| 11:35.29 | _zoom_ | ok |
| 11:37.42 | _zoom_ | and you handle all refreshing stuff in your plugin, mean info updates |
| 11:37.56 | _zoom_ | or you just let it to admin |
| 11:39.16 | fenrus | i guess ill parse 'sip show registry |
| 11:39.18 | fenrus | ' |
| 11:39.40 | fenrus | and trigger if registered isnt on all lines |
| 11:46.11 | Cain | How many ip-phones/soft-phones can be handled by an asterisk server ? (PowerEdge 830 Server with Pentium D@3.00Ghz/2GB Ram) |
| 11:46.39 | Cain | (TDM410P with 4 FXO modules inside) |
| 11:47.55 | Tozz_ | lots. |
| 11:48.09 | Cain | 100 or so ? |
| 11:48.24 | Tozz_ | much more, but it depends if u do codec translation |
| 11:48.28 | Tozz_ | codec translation costs CPU |
| 11:48.38 | Tozz_ | if u dont do codec translation u can have thousands |
| 11:49.12 | Tozz_ | but also depends on if ur running dialplan scripts that are CPU intensive |
| 11:49.21 | Tozz_ | if u use Mysql and what not |
| 11:50.32 | Cain | Tozz_ : okey, so it's just a matter of optmization after ^^ |
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| 11:56.34 | OldMonk | hi. anyone help me debug this 4-line dialplan? error is: 12:30:09] NOTICE[1762]: chan_sip.c:20152 handle_request_invite: Call from '3000' to extension '09811066460' rejected because extension not found in context 'from-internal' |
| 11:56.51 | OldMonk | http://pastebin.com/wPpNRJdD |
| 11:57.41 | OldMonk | if i change the order of the lines (first 2 lines later, 3rd and 4th lines first), external calls work but internal extensions don't |
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| 12:08.02 | OldMonk | ok, i figured it out |
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| 15:10.36 | jamko | Morning. How is it that some attackers are getting the user names of my UA? Does asterisk give a specific response when only the password is wrong? |
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| 15:37.57 | pabelanger | jamko: what is an example of a username? |
| 15:38.30 | jamko | Well just as an example.. rof101 |
| 15:39.14 | pabelanger | jamko: Ya, that be pretty easy to bruteforce, why not use mac address as usernames? |
| 15:40.55 | jamko | pabelanger: Well my question is more of a concern of how they are getting it. My passwords are normally over 15 characters and not dictionary friendly. I was just wondering if asterisk is providing a response that |
| 15:41.16 | pabelanger | jamko: Not asterisk, SIP |
| 15:41.41 | jamko | of wrong pass, correct user. |
| 15:41.43 | jamko | Ok |
| 15:42.14 | jamko | Or if somehow a remote attacker was sniffing my traffic, which from what I understand is not possible unless BGP is compromised through my isp. |
| 15:43.01 | *** join/#asterisk thews (~thews@24-119-172-234.cpe.cableone.net) |
| 15:44.13 | pabelanger | jamko: So, with SIP if there is no user Asterisk would return 404 Not Found, but if it did exist, it would reply 401 Unauthorized (then start authentication) |
| 15:45.45 | WIMPy | see alwaysauthreject= in sip.conf |
| 15:46.21 | jamko | thanks |
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| 15:55.22 | jamko | Sometimes though it just seems as if attackers are sniffing traffic on my box. For example, I will see the same attacker simultaneously going after two of my boxes, that are on opposite sides of the US. Logically I would think this is because |
| 15:55.44 | jamko | they were sniffing where traffic was being sent to from the phones. |
| 15:55.57 | jamko | But shit what do I know.. : ) |
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| 16:50.01 | DelphiWorld | hello |
| 16:50.06 | DelphiWorld | anyone using asterisk + MGCP? |
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| 17:06.04 | _zoom_ | DelphiWorld: hello |
| 17:06.17 | DelphiWorld | hello brother _zoom_ |
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| 17:29.18 | DelphiWorld | someone please help me configuring asterisk for mgcp |
| 17:35.37 | DelphiWorld | how do i list loaded modules? |
| 17:44.44 | erinspice | core module show? |
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| 18:07.44 | mechbangirc | hi intel IPP codecs, are they passthrough only? |
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| 18:23.45 | mactimes | #elastix |
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| 18:35.10 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 18:54.29 | DelphiWorld | hi pabelanger |
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| 19:00.03 | DelphiWorld | i put chan_sip.so in the module.conf but is unable to load... any idea? |
| 19:07.00 | ChannelZ | Does it say why? do you have a valid sip.conf? |
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| 19:26.32 | DelphiWorld | ChannelZ: got the issue. |
| 19:26.41 | DelphiWorld | ganwads some simbole issue... recompiling. |
| 19:27.49 | ChannelZ | I have no idea what that means but OK |
| 19:28.18 | DelphiWorld | ChannelZ: i use openwrt |
| 19:28.41 | DelphiWorld | ChannelZ: i built it myself but got the asterisk package from the repo... so didn't work, while recompile it. |
| 19:29.00 | ChannelZ | I meant the 'ganwads some simbole' |
| 19:39.41 | DelphiWorld | ChannelZ: ganwads is a typing error :D |
| 19:51.20 | DelphiWorld | ChannelZ: have you evver used mgcp? |
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| 20:28.02 | gaccardo | Hello, is posible to get connected peers using mysql or put a register in the database when user connects?... i have asterisk-addons working or |
| 20:31.27 | ChannelZ | DelphiWorld: nope |
| 20:35.36 | DelphiWorld | ChannelZ: nm nm nm |
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| 20:53.02 | *** join/#asterisk ixti (~ixti@217.70.249.69) |
| 20:53.10 | ixti | Hello everybody |
| 20:53.57 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 20:54.07 | ixti | Can somebody help with SendFax() I can't get it to work. ReceiveFax() work fine, while SendFax() not. I'm trying res_fax digium. |
| 20:54.37 | ixti | When I'm starting SendFax() I see lots of errors in console: UDPTL asked to send 59 bytes of IFP when far end only prepared to accept 30 bytes; data loss will occur.You may need to |
| 20:54.47 | ixti | override the T38FaxMaxDatagram value for this endpoint in the channel driver configuration. |
| 20:55.13 | ixti | I have already wrote into my sip.conf: t38pt_udptl=yes,fec,maxdatagram=30 |
| 20:55.48 | ixti | But error still exists. Fax starts, but then lots of errors comes to console, and "receiving error" appears on remote fax machine :(( |
| 21:28.30 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 21:29.23 | nextime | hello all. Is skype for asterisk from digium supporting video ( meaning, can i make a videocall between a sip client and a skype client using it in my asterisk server? ) |
| 21:31.25 | DelphiWorld | nextime: in my understanding, only video and maybe text... text is not 100% sure. |
| 21:32.12 | nextime | DelphiWorld : text doesn't matter for my needs, just audio/video calls |
| 21:32.27 | DelphiWorld | ok, nextime, so no video for now :-) |
| 21:33.19 | nextime | ok, thanks |
| 21:33.45 | DelphiWorld | nextime: np |
| 21:42.36 | ixti | can somebody help? |
| 21:43.00 | ixti | any guess will be appreciated |
| 21:43.00 | *** join/#asterisk OnTheRock (~IceChat77@209.240.42.73) |
| 21:43.19 | ixti | because i'm totally lost |
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| 21:46.27 | DelphiWorld | night people and good weekend |
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| 22:37.15 | a1fa | hello, i need to rebuild my dial plan.. it's getting out of hand; is there a good framework to use? ie, that's going to check if extension is busy, available, etc and such |
| 22:54.53 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 23:10.11 | a1fa | what is the command to look at dialplan as call comes in? |
| 23:26.55 | a1fa | anyway to simulate "beep-beep" for Waitexten()? :) |
| 23:26.57 | ChannelZ | asterisk -rv |
| 23:27.18 | ChannelZ | beep-beep? |
| 23:33.15 | raden | p3nguin, you around bro ? |
| 23:33.48 | a1fa | Playtones() thing.. but cant figure out how to make any sound |
| 23:34.58 | a1fa | nothing plays with Playtones |
| 23:35.10 | a1fa | so Playtones(ring) |
| 23:35.11 | a1fa | forexample |
| 23:39.03 | a1fa | hm.. it worked for dial |
| 23:39.06 | a1fa | strange :) |
| 23:51.47 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 23:52.52 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |