| 00:00.22 | p3nguin | But they might soon, so buy buy buy! |
| 00:00.43 | WIMPy | I see, you got it. |
| 00:01.29 | WIMPy | We could try to sell IPv7 addresses of future insecure mailservers. |
| 00:02.49 | *** part/#asterisk fireman_biff (~biff@65.48.133.102) |
| 00:03.30 | telnettech_lap | can someone explain hoe the include => statement works? if I have a statement for the local calling context included in the Long distance context, If I included the long distance context in the international context, will the local context aslo part of international context? |
| 00:03.39 | telnettech_lap | how |
| 00:04.04 | WIMPy | Sure |
| 00:04.26 | WIMPy | You include the whole context. Doesnt matter where it's coming from. |
| 00:04.42 | p3nguin | If you include context2 inside context1, calls landing in context1 will have access to extensions in context2. |
| 00:05.08 | telnettech_lap | ok thanks....thats what i thought but wanted to make sure....dont want the kids to be able to dial internationally |
| 00:05.33 | p3nguin | Dialplan should be built hierarchically for that purpose. |
| 00:07.54 | p3nguin | Do you understand what I mean by that? Do you need an example? |
| 00:08.37 | ManxPower | WIMPy, riddlebox remove the secret= line for an account they are trying (or create one), route all calls to a message saying the person is using a stolen phone card and the FBI and INS will be sent their phone number. Took about 45 mins and the calls stopped |
| 00:08.43 | ManxPower | they have not been back in 2 days |
| 00:09.30 | WIMPy | I didn't have anyone trying to place calls. |
| 00:09.31 | ManxPower | based on chanspy it sounded like people might have bought calling cards to call home or something. |
| 00:09.42 | WIMPy | I just had some attacks trying to send faxes. |
| 00:09.51 | ManxPower | Ah. You will be attacked. |
| 00:10.15 | ManxPower | XO sent us a notice telling us about a significant increase in fraud by hacked PBXs. |
| 00:10.58 | riddlebox | ManxPower, I have thought about doing that |
| 00:11.34 | ManxPower | if nothing else whoever sold them access will have some pissed off customers |
| 00:12.26 | riddlebox | fail2ban, and some iptables rules keep everyone at bay, but I didnt have alwaysauthreject=yes |
| 00:12.33 | WIMPy | Do you think they send customer calls unless a (probably automatic) test confirms that it's working? |
| 00:13.14 | telnettech_lap | p3guin: no I understand....thats why I was asking the question so that I dont give someone access that should have certain rights |
| 00:14.00 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 00:14.20 | ManxPower | WIMPy, They had found an account with a bad password, so calls were already flowing. I imagine allowing a few calls thru first would make it more effective |
| 00:14.40 | WIMPy | yes, ok |
| 00:15.00 | ManxPower | the customer manages his own pbx. |
| 00:15.45 | lirakis | sounds like a bad idea lol |
| 00:16.02 | ManxPower | lirakis, we discourage the practice |
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| 00:17.53 | p3nguin | The only good thing about that is that you aren't responsible for that compromise. |
| 00:18.13 | leifmadsen | unlimitel has also seen the same increase in fraud. I imagine most, if not all phone companies have lately |
| 00:18.36 | leifmadsen | if you see any ISP or provider selling super cheap minutes, it's likely a re-sell scam of hacked systems |
| 00:19.03 | leifmadsen | on an upside, my fail2ban works so well I locked myself out after reconfiguring my phone incorrectly and not updating my ignoreip= field in jail.conf |
| 00:19.41 | leifmadsen | would really enjoy setting up a honeypot and monitoring what happens :) |
| 00:19.57 | WIMPy | Ja, that's what florz always points out. You just replace one risk with another. |
| 00:19.59 | leifmadsen | then screwing with people who think they have access to a hacked system. Much like ManxPower did with the message about the FBI and INS :) |
| 00:21.23 | riddlebox | leifmadsen, play screaming monkey's over and over when they try to dial anything |
| 00:23.12 | telnettech_lap | so is BLACKLIST as an application removed from Asterisk 1.8 ? |
| 00:23.43 | p3nguin | it's a function |
| 00:24.18 | Baylink | QUERY: do mod_rpt and other radio over IP folks hang out here? Or somewhere else? |
| 00:25.13 | telnettech_lap | so how do you call it so that you can check the callerID to see if the person is on the Blacklist in the DB |
| 00:25.33 | Juggie | pats leifmadsen on the back :) |
| 00:25.37 | p3nguin | GotoIf($[${BLACKLIST()}]?misc,blocked,1) |
| 00:26.03 | p3nguin | exten => blocked,1,Playback(silence/1&privacy-you-are-blacklisted&vm-goodbye) |
| 00:30.43 | *** join/#asterisk gazzerh (~garryh@93-97-187-150.zone5.bethere.co.uk) |
| 00:35.45 | ManxPower | reading UPGRADE*.txt in the asterisk source dir is a good place to find out information about your version of Asterisk |
| 00:36.05 | ManxPower | It also helps you translate the docs you find online into your version of Asterisk |
| 00:37.45 | AAronCI | sigh, I've installed asterisk 1.6 and it doesn't even start up :( |
| 00:37.52 | AAronCI | keeps giving me this error "Unable to connect to remote asterisk (does /opt/var/run/asterisk/asterisk.ctl exist?)" |
| 00:38.58 | pabelanger | AAronCI: What is the output from asterisk -vvvvvcg |
| 00:38.59 | WIMPy | That mens you cannot connect to a running asterisk. |
| 00:39.08 | WIMPy | Have you tried starting asterisk with -c? |
| 00:39.50 | AAronCI | I'm just going to quickly reboot |
| 00:41.45 | AAronCI | asterisk -vvvvvcg gives me lots of text, but the last line is a segmentation fault |
| 00:42.02 | p3nguin | Sounds like some type of lib mismatch. |
| 00:42.24 | pabelanger | AAronCI: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 00:44.42 | AAronCI | there's no core files in /tmp. In fact, there aren't any core files anywhere on my system. |
| 00:45.31 | AAronCI | asterisk was installed from ipkg and was built for an embedded system. Maybe it was optimized and can't create core dumps |
| 00:45.59 | WIMPy | Maybe you could disable automatic module loading and see how far you get trying to load them manually. |
| 00:46.41 | AAronCI | hmm, not a bad idea. |
| 00:46.54 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 00:46.54 | AAronCI | asterisk won't start without any modules right? |
| 00:47.14 | AAronCI | er, won't start with no modules loaded |
| 00:47.15 | WIMPy | It will start, but it won;t do much. |
| 00:47.21 | AAronCI | oh, ok |
| 00:47.24 | AAronCI | i'll try that |
| 00:48.18 | AAronCI | alright, made modules.conf a blank file |
| 00:48.27 | AAronCI | see what happens now |
| 00:49.23 | WIMPy | Actually autoload=no might be needed. |
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| 00:49.49 | AAronCI | yeah, right now it's telling me there is no command called sip |
| 00:50.06 | WIMPy | That's perfectely normal. |
| 00:51.15 | WIMPy | Maybe you should take a look through the modules and get rid of those you won't need anyway. |
| 00:52.03 | AAronCI | well I've got only autoload=no in the file and it _seems_ to be fine. It says 'asterisk ready' |
| 00:52.29 | p3nguin | Doesn't modules.conf require at least [modules] in it? |
| 00:52.40 | WIMPy | Yes, but it's not very usefull now. |
| 00:52.51 | AAronCI | I put it in |
| 00:53.09 | pabelanger | AAronCI: PB the output from asterisk -vvvvvcg when it segfaults. |
| 00:53.16 | pabelanger | which version of asterisk? |
| 00:54.14 | AAronCI | it's Asterisk 1.6.2.13. But it's not segfaulting now that I have no modules loaded |
| 00:54.39 | p3nguin | Now you can start manually loading the ones you do need. |
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| 00:55.46 | smellis | has anyone been able to pair an iphone to a centos 5.5 server for the purposes of using chan_mobile with asterisk? |
| 01:00.36 | AAronCI | Ok, well this is going well so far, but i'm so tired I'm going to put this on hold and come back tomorrow. |
| 01:00.47 | AAronCI | I'd like to thank everybody for the help that they have give though |
| 01:00.57 | AAronCI | It is much appreciated by somebody who is very new at this |
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| 01:24.21 | telnettech_lap | goodnight guys....talk at ya tomorrow |
| 01:33.22 | p3nguin | Hmm, I found a bug in chan_sccp-b v3. When using CFwdAll, the phone does not display that the phone is in forwarding mode. |
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| 01:39.11 | p3nguin | Anyone else using chan_sccp-b v3 and you do see the forward on the display? |
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| 01:51.34 | p3nguin | I guess there is already some discussion about this problem in the bugtracker for the project. Maybe they will get it fixed soon. |
| 02:10.00 | p3nguin | Well, the latest svn build seems to address that problem. It's not perfect, but it's better than not knowing at all that the phone is forwarding. |
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| 02:45.46 | sshock | Hi, let's say I have several extensions that can be called directly (sip:NUM@example.com) |
| 02:46.34 | sshock | what's the easiest way to put in some code to set the incoming callerid ? |
| 02:47.33 | sshock | have a macro that does it, and then on every extension call that macro first thing before dialing the sip user? |
| 02:48.04 | sshock | having a macro simplifies things, but I still have to call it from several places... is there an easier way? |
| 02:48.54 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 02:49.11 | p3nguin | What do you want to set the incoming caller id to? |
| 02:51.16 | sshock | actually it's the caller id name |
| 02:51.31 | p3nguin | You don't want to accept what the caller sends? |
| 02:51.31 | sshock | I already have a perl AGI set up that sets the caller id name from the caller id number. |
| 02:51.48 | sshock | no, at the moment I'm not getting any CNAMs at all |
| 02:52.59 | sshock | is there some way to execute stuff at the beginning before it calls the extension? |
| 02:53.16 | sshock | I can't use the "s" extension, because it doesn't go through that when sip user is called directly |
| 02:53.22 | p3nguin | Before a call is placed? Of course not. |
| 02:53.55 | sshock | p3nguin: not before a call is placed; when a call is received |
| 02:54.21 | p3nguin | <sshock> is there some way to execute stuff at the beginning before it calls the extension? |
| 02:54.50 | p3nguin | When the extension is called, that's when the call begins. Set the caller ID name early in the dialplan. |
| 02:55.13 | p3nguin | Before dialing the phone, of course. |
| 02:55.35 | sshock | p3nguin: I can solve it just fine, but I don't want to have to put Macro(setcallerid) as the first step of every extension |
| 02:55.58 | p3nguin | Macros are deprecated anyway. |
| 02:56.11 | p3nguin | Gosub/Return is the new way. |
| 02:56.13 | sshock | oh; that is good to know |
| 02:56.31 | sshock | in any case, I guess it is still the same issue |
| 02:57.22 | p3nguin | You can always use patterns to reduce the number of extensions. |
| 02:57.22 | sshock | if I have 500 extensions, I'd like to have only one step for each extension, not two steps |
| 02:57.30 | sshock | patterns? |
| 02:59.25 | p3nguin | exten => _[245]XX,1,Verbose(this matches 200-299 and 400-599) |
| 02:59.33 | sshock | oh, those |
| 02:59.48 | sshock | hmm, I'm not sure that actually solves this case, does it? |
| 03:00.12 | sshock | unless... |
| 03:00.39 | sshock | hmm, it might just work; ok, I'll try it now... |
| 03:00.41 | p3nguin | You'll still have to set the caller id, but you'll reduce the number of extensions. |
| 03:01.14 | p3nguin | Instead of having 100 individual extensions, you could have 1. |
| 03:01.46 | sshock | and after doing my thing (set the caller name), then I can jump to another (the real) extension |
| 03:01.57 | p3nguin | It depends on other factors if it will work in your exact case, though. |
| 03:02.16 | sshock | e.g., 200-299 just sets the caller Id and then calls exten+100 (300-399) which does the actual dial |
| 03:02.58 | sshock | hmm, worth a try; I can write like $[${EXTEN} + 100] right? |
| 03:03.21 | p3nguin | Yeah, you can match everything, set caller id, then Goto() the new place. |
| 03:03.43 | sshock | cool, I think this will actually work; too bad I have to use up two ranges though... :( |
| 03:04.21 | p3nguin | _X!,1,Goto(other place) |
| 03:04.25 | sshock | I wish there was like a "context-begin" extension or something |
| 03:04.46 | sshock | hmm, actually maybe I don't need two ranges |
| 03:05.01 | p3nguin | _X! might not be the best way, but it could work. |
| 03:05.02 | sshock | since * will match the more specific extension, right? |
| 03:05.09 | p3nguin | yes |
| 03:05.13 | sshock | yeah, so it won't call _X! again |
| 03:05.35 | p3nguin | _X. might be better |
| 03:05.46 | p3nguin | as long as you have at least two numbers. |
| 03:05.56 | sshock | yeah, my extension are all 3 digits long |
| 03:06.10 | sshock | maybe I should do _XXX then |
| 03:07.00 | p3nguin | At least you've got something to play with for a while. |
| 03:07.55 | sshock | yeah, I think this is going to work, as long as the Goto() doesn't jump back to the _XXX rule and cause * to hang :) |
| 03:08.23 | sshock | hmm, wait a sec; that's not going to work |
| 03:08.35 | p3nguin | If you have specific extens, it should be fine. |
| 03:08.47 | sshock | because * will never pick _XXX in the first place; if someone dials 100@example.com, it will go straight to 100; it's not going to do _XXX first |
| 03:08.56 | sshock | I'll have to do the two ranges |
| 03:09.21 | p3nguin | Right, but you have to use a context in between. |
| 03:09.39 | sshock | _1XX,1,Macro(setcallerid) _1XX,n,Goto($[${EXTEN} + 100]) |
| 03:09.49 | sshock | what do you mean a context in between? |
| 03:10.01 | p3nguin | no |
| 03:10.42 | sshock | hmm, just had another idea; instead of adding 100, I can just prepend something |
| 03:11.02 | sshock | e.g., _1XX,n,Goto(real${EXTEN}) |
| 03:11.23 | sshock | then I all my real extensions would be like real100, real101, etc. |
| 03:12.36 | sshock | of course, then if anyone ever figures out what I'm doing, they could call sip:real100@example.com directly and bypass my setcallerid |
| 03:12.50 | p3nguin | http://pastebin.com/5d2DDJL0 |
| 03:13.08 | sshock | ooohhh, duh |
| 03:13.21 | sshock | just use two contexts; I gotcha |
| 03:14.09 | sshock | adding another context always solves everything :) |
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| 03:14.13 | p3nguin | Your suggestion of masking the exten would also work and it would allow you to use a single context. |
| 03:14.13 | sshock | thanks |
| 03:14.28 | sshock | I think I like the two contexts better though |
| 03:15.08 | p3nguin | http://pastebin.com/mydGdU17 |
| 03:15.37 | sshock | yep |
| 03:15.45 | p3nguin | Either way should work. |
| 03:15.57 | sshock | hmm, now suppose I also have usernames that can be called directly, like sip:john@example.com |
| 03:16.18 | sshock | what extension could I use to match anything? I'm gussing _X! won't work because the first character would have to be a digit |
| 03:16.46 | sshock | hmm, the . (period) matches anything, but the book says to be careful about using that |
| 03:17.17 | sshock | it says never use _. |
| 03:17.24 | p3nguin | What about [a-z] ? |
| 03:17.45 | sshock | hmm, good idea; does that actually work? |
| 03:18.06 | p3nguin | exten => _[a-z].,1,Verbose(try it) |
| 03:18.22 | p3nguin | I have no idea if it will work or not. |
| 03:19.00 | sshock | ok, I'm gonna try it now... |
| 03:19.33 | p3nguin | I don't see why it wouldn't work, but I've never used that pattern for anything. |
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| 03:19.56 | p3nguin | a to z, plus at least one more character (number or letter) |
| 03:20.31 | p3nguin | If you need caps, you'd have to create another exten using _[A-Z]. too. |
| 03:20.39 | sshock | ok |
| 03:21.36 | p3nguin | Now you're up to three extens to capture all calls. |
| 03:23.07 | p3nguin | Still better than adding a Set() on all extensions. |
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| 03:24.26 | sshock | yep; seems to be working |
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| 04:14.01 | nny | if you have to choose between wav or WAV for recording, any preferences? I know one is raw wav, playable by almost anything and the other is the compressed wav, not playable by iphones (iirc) etc |
| 04:19.26 | ChannelZ | depends on your use |
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| 04:21.22 | ChannelZ | I'd say regular wav is preferable in general |
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| 05:00.21 | sshock | so if ENUM took over the world, then we could all start calling each other for free? |
| 05:01.26 | sshock | and PSTN could be obselete? that would be awesome! |
| 05:05.21 | sshock | so do I have to pay anyone to get a number in the e164.arpa dns? |
| 05:08.56 | sshock | cool, looks like e164.org will let you add your PSTN number for free, though donations are welcome |
| 05:12.01 | sshock | hmm, but e164.org is not the same as the official e164.arpa; I wonder what it takes to get added to the official one |
| 05:14.26 | drmessano | I think ENUM is a waste of time |
| 05:15.07 | drmessano | We should be using URI's and not some throwback to yesteryear |
| 05:16.10 | sshock | well, I think it makes sense for helping to converge the old with the new |
| 05:16.46 | sshock | most people still have "normal" phones and are used to dialing numbers I imagine |
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| 05:40.38 | sshock | sounds like the official ENUM for country code 1 (1.e164.arpa) is still in the planning stages: http://www.enumllc.com/ |
| 05:40.52 | tash | does anyone in here have experience configuring Asterisk with an E1 in Greece? |
| 05:41.49 | tash | I'm having a hell of a time with this E1 config |
| 05:41.55 | sshock | I wonder how much the PSTN telcos are paying them to take their sweet old time... |
| 05:42.03 | tash | either someting I'm doing wrong or the telco isn't providing good enough info |
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| 05:49.57 | boodu | bye |
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| 05:53.54 | _omer | hello |
| 05:54.23 | _omer | I need suggestion to make it possible ? .... http://www.pastebin.ca/1980856 <--- please check |
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| 06:37.01 | atan | Well seeing Goog-411 is going out of style, are there any other decent 411 providers? |
| 06:37.10 | atan | Or perhaps even paid ones somehow? |
| 06:43.39 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 06:49.25 | ChannelZ | 911 |
| 06:52.42 | atan | ChannelZ, yes, I'll forward my callers to 911 for their pizza order. Yes. Perfect. |
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| 06:54.26 | atan | I sense a market for 411 service. |
| 06:54.49 | atan | Except I wonder how many databases one could use :P |
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| 07:12.47 | kaldemar | _omer: remove any timeouts |
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| 07:47.58 | schmidts | good morning |
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| 08:02.15 | atan | schmidts, morning |
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| 08:16.17 | *** join/#asterisk timnis (~timnis@YZMMCCCXXXIV.gprs.sl-laajakaista.fi) |
| 08:18.30 | timnis | Hi, I have try to setup TLS client but I get "tcptls.c: FILE * open failed!" when I try to call TLS client |
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| 08:43.13 | skyion | has anyone has experience with streaks in faxes received within asterisk? |
| 08:44.43 | timnis | I have installed asterisk 1.8 and configured TLS. Registering from TLS is ok but when I try to call to TLS registered phone I get error msg "FILE * open failed!"? |
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| 08:57.07 | *** join/#asterisk burnz84 (~burnz84@212.80.243.19) |
| 08:57.18 | burnz84 | good morning |
| 08:57.21 | burnz84 | is it possible to set on char before the calleridnum? for example: some one calls the general office an there is no one, after 20 seconds ringing there is a forwading to an other employee. And he must see that the caller first have call the general office. i want to display it like C(for cantral)XXXXXXX(x for incoming caller number) at the telephon display. |
| 08:57.30 | *** join/#asterisk anny__ (~chatzilla@77.42.241.114) |
| 09:01.00 | kaldemar | burnz84: yes, put a Set(CALLERID(num)=C${CALLERID(num)}) in your dialplan |
| 09:01.58 | burnz84 | thank you |
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| 09:16.02 | burnz84 | mhh it doesen´t work. i have placed it at the first line to test it. |
| 09:19.15 | kaldemar | show the extension and CLI output of a call |
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| 09:45.40 | *** join/#asterisk gregd (~gregd@188-220-38-34.zone11.bethere.co.uk) |
| 09:46.50 | gregd | hi guys, is there any 'magic' with setting up a nokia sip with asterisknow? I can connect to asterisk with any other sip softphone.. but when trying to connect with nokia the following comes up in the log: username mismatch, have <101>, digest has <6000> |
| 09:47.28 | gregd | could someone enlighten me please? |
| 09:48.04 | *** join/#asterisk Razr3J (~Razr3J@112.201.253.133) |
| 09:48.17 | Razr3J | hello! |
| 09:48.53 | Montys | @gregd could you please show me the complete line of the error/warning/notice? |
| 09:49.16 | gregd | Montys: sure, i will paste it somewhere, one moment pls |
| 09:49.46 | gregd | Montys: http://pastebin.com/XtJHvR9z |
| 09:50.48 | Montys | what is the user name that you set in Asterisk for the phone? |
| 09:51.05 | Montys | or "extension" |
| 09:51.08 | gregd | Montys: the extension is 101 |
| 09:51.37 | burnz84 | now it works@kaldemar...thank you very much |
| 09:52.04 | Montys | gregd, Could you please confirm if sip.cobf or user.conf has that extention? |
| 09:52.24 | Montys | etc/asterisk/sip.conf or /etc/asterisk/users.conf |
| 09:52.26 | gregd | Montys: yes, the sip_additional.conf has it |
| 09:52.35 | gregd | Montys: but this is AFAIK incuded from sip.cofn |
| 09:53.00 | Montys | as long you have #include sip_additional.conf in sip.conf it should work |
| 09:53.07 | gregd | anyway, I've got other extensions configured.. and they work. .moreover, the 101 extension works when registering with xlite or any other softphone |
| 09:53.28 | gregd | just does not work with the nokia sip client |
| 09:54.55 | Montys | @gregd could you paste the portion of sip_aditional where you setup that extention? |
| 09:54.58 | Montys | extension? |
| 09:57.24 | gregd | Montys: sure, one moment pls |
| 10:00.03 | gregd | Montys: http://pastebin.com/GXQr5b1D |
| 10:03.19 | kaldemar | gregd: the nokia client is authenticating by name 6000. |
| 10:03.52 | Montys | @gregd; I changed your configuration |
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| 10:04.37 | Montys | Kaldermar; yeah that could be too; I was going to suggest check the nokia phone, but I want to make sure about * 1st |
| 10:04.40 | gregd | kaldemar: why is it authenticating as 6000? |
| 10:05.16 | kaldemar | gregd: it's probably told to do so. check the configuration of the phone. |
| 10:05.48 | gregd | ok, let me restart the phone |
| 10:16.03 | gregd | still got the same issue... the nokia phone authenticates as 6000 :/ |
| 10:16.14 | gregd | I'll try creating a new profile from scratch |
| 10:21.35 | *** join/#asterisk [netman] (~netman@25.Red-79-156-228.staticIP.rima-tde.net) |
| 10:24.26 | kjs | hmm, I am getting no audio when dialing out of my asterisk box, yet inbound calls are fine... |
| 10:25.21 | gregd | Montys: I'm sure I the user is defined to be 101 on the phone... why do you think it is authenticating as 6000? |
| 10:25.37 | gregd | is there still other way of finding it out except of debugging the protocol? |
| 10:27.50 | kaldemar | gregd: "digest has <6000>". enable sip debug and pastebin a CLI output. |
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| 10:43.25 | Razr3J | hellp |
| 10:46.24 | kjs | Hmmm could this error message be related to no outbound audio in either direction? http://pastebin.ca/1982663 |
| 10:47.56 | *** join/#asterisk henk (~henk@netwichtig.de) |
| 10:47.58 | henk | hi |
| 10:48.21 | henk | my server is being flooded by registers which raises the load and makes it a bit unusable. is there a sane way to protect against that? |
| 10:48.32 | kaldemar | kjs: sure. see this: |
| 10:48.34 | kaldemar | ~sipnat |
| 10:48.35 | infobot | i guess sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 10:49.05 | kaldemar | henk: drop the ip range(s) with iptables. |
| 10:49.32 | *** join/#asterisk joel_oliveira (~chatzilla@alpes.nortenet.pt) |
| 10:49.45 | henk | kaldemar: that's not sane. |
| 10:51.36 | henk | kaldemar: well, it might be sane, but definitely not a solution... there has to be an attack already to know what to block and i have to manuelly block it. |
| 10:51.45 | kjs | kaldemar: people have refered me to this before however, the link is dead. |
| 10:52.24 | kaldemar | henk: people use fail2ban to automatically block malicious addresses. |
| 10:52.37 | henk | kaldemar: i am not around 24/7 and the server's load goes beyond the point the server still reacts in a few minutes... |
| 10:53.21 | henk | kaldemar: never liked that with ssh and don't like it for asterisk any better. do you know of anything else? |
| 10:54.07 | kaldemar | henk: sorry, no. |
| 10:54.40 | henk | kaldemar: ok, thank you :) |
| 10:54.49 | kaldemar | kjs: if there's a NAT involved, see the sample sip.conf for information on the NAT parameters. |
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| 11:06.35 | gregd | kaldemar: ok, here is my full sip debug for the authentication part... http://pastebin.com/48nSFVcm |
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| 11:06.46 | gregd | could you please take a look at it? |
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| 11:13.39 | fauxalliance | gregd, any crazy firewalls? |
| 11:14.03 | gregd | fauxalliance: no, i dont think so... im on the same subnet |
| 11:14.30 | fauxalliance | gregd, and you are _sure_ the credentials are good... line 100 barks SIP 403 |
| 11:15.36 | gregd | fauxalliance: what do you mean? I try to connect via extension 101, not 100 |
| 11:16.03 | fauxalliance | gregd, 100. SIP/2.0 403 Forbidden (Bad auth) |
| 11:16.13 | fauxalliance | ^^LINE 100 in your pastebin |
| 11:16.38 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 11:17.19 | kaldemar | gregd: the nokia definitely sends "username="6000" in the digest authentication. |
| 11:17.30 | gregd | yes I am sure that the credentials are fine... |
| 11:17.43 | fauxalliance | ^^sure? |
| 11:18.07 | kaldemar | they clearly are not fine because asterisk expects 101 and the phone sends 6000. |
| 11:18.16 | fauxalliance | thank you kaldemar |
| 11:18.18 | fauxalliance | kaldemar: +1 |
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| 11:58.55 | gregd | how do I change realm in asterisknow from 'asterisk' to something else? |
| 11:59.03 | gregd | is it possible at all? |
| 12:01.13 | russellb | it's a sip.conf option |
| 12:01.20 | russellb | if you're using FreePBX, i have no idea if it lets you change it |
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| 12:08.51 | gregd | great.. |
| 12:09.18 | gregd | fauxalliance: i solved my problem with the mismatch by setting realm in [general] sip.confg |
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| 12:25.32 | timnis | Have anyone setup asterisk with TLS SIP? I have little problem with it I can register but I cannot make a call... |
| 12:25.47 | *** join/#asterisk boch (c829e3f5@gateway/web/freenode/ip.200.41.227.245) |
| 12:25.56 | boch | hi all |
| 12:29.26 | underdog | . |
| 12:30.13 | *** part/#asterisk skyion (~bradc@siza.thusa.net) |
| 12:33.50 | *** join/#asterisk SpaceBass (~SP@pool-98-117-75-200.rcmdva.fios.verizon.net) |
| 12:33.58 | SpaceBass | hey folks |
| 12:34.20 | SpaceBass | any google voice users? My dtfm tones just stopped working with any GV calls - verifying new numbers, accpeting calls, etc |
| 12:35.52 | russellb | using which version of asterisk? |
| 12:36.01 | russellb | i wonder if they shut us down, heh |
| 12:36.43 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
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| 12:37.15 | SpaceBass | russellb, its strange, quite literally happened without any changes on my end - using 1.8.0 |
| 12:37.40 | russellb | so calls are going through, just no more DTMF? |
| 12:37.47 | SpaceBass | im also having dftm problems at my office, some kind of proprietary PBX, so I'm leaning towards this being a DTFM issue |
| 12:38.05 | SpaceBass | russellb, yeah, i can recieve calls on my gizmo trunk just fine, but I can't answer them b/c I cant send the 1 |
| 12:38.21 | russellb | oh, you're doing it through gizmo? |
| 12:38.27 | russellb | 1.8.0 has native googlevoice support now :-) |
| 12:38.47 | russellb | you should try it |
| 12:38.50 | russellb | ~googlevoice |
| 12:38.50 | infobot | For information on setting up calls with google using Asterisk 1.8, see https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 12:38.54 | SpaceBass | russellb, I'd gladly move to the native support, but for the life of me I couldnt find the docs on it |
| 12:38.55 | SpaceBass | thanks |
| 12:38.59 | russellb | np |
| 12:41.11 | *** join/#asterisk some0ne (some0ne@41.205.43.107) |
| 12:41.48 | some0ne | can anyone tell me a good site that teaches hoe to resolve the sip/nat issue? |
| 12:41.56 | russellb | ~sipnat |
| 12:41.56 | infobot | sipnat is probably Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 12:42.10 | some0ne | thnx!! |
| 12:42.14 | pabelanger | woah, SIP/2.0 101 Dialog Establishement |
| 12:42.21 | some0ne | that was quick |
| 12:42.26 | russellb | pabelanger: that's a new one on me |
| 12:42.28 | russellb | some0ne: :-) |
| 12:42.28 | pabelanger | don't think I have seen that before |
| 12:42.53 | some0ne | thanks lot russellb |
| 12:43.02 | russellb | some0ne: good luck to ya |
| 12:45.27 | some0ne | did anyone here were able to resolve it? |
| 12:45.49 | some0ne | not sure weather I use a tunnel |
| 12:45.58 | some0ne | or iptbles |
| 12:46.32 | some0ne | (netfilter) |
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| 12:56.22 | Phlog | Hi, anybody any idea whether ZRTP passthrough made it into asterisk 1.8?? |
| 12:57.23 | ManxPower | Phlog, All significant changes are documented in the UPGRADE*.txt files included with Asterisk |
| 12:59.00 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 12:59.39 | SpaceBass | drat, setting up GV on a box with freepbx is a PITA |
| 13:00.21 | ManxPower | SpaceBass, at least there is a channel for you, #freepbx |
| 13:01.11 | SpaceBass | yeah, Im there :D |
| 13:01.16 | *** join/#asterisk bmint (~bmint@h174.92.190.173.static.ip.windstream.net) |
| 13:04.41 | jamko | Morning... Has anyone used both Voip.ms and Flowroute for origination? Other than pricing, any positive / negative feedback on the quality of service would be great. Thanks.. |
| 13:07.01 | [TK]D-Fender | jamko: We've have several Flowroute issues float through here lately, voip.ms seems pretty stable throughout |
| 13:07.25 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 13:07.38 | jamko | [TK]D-Fender: Thanks.. What type of issues? |
| 13:07.49 | ManxPower | isn't voice.ms just a vitelity reseller? |
| 13:07.56 | ManxPower | ..er.. voip.ms |
| 13:08.14 | [TK]D-Fender | jamko: I dont really recall... general stuff like timeouts ETC I believe... |
| 13:08.17 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:08.48 | Katty | HAI GUYS |
| 13:09.10 | jamko | ManxPower: would explain why their rates are generally on the higher end of the spectrum. |
| 13:10.04 | ManxPower | Much like cell phone service, most of the differences between VoIP providers is customer service and coverage. I personally use Vitelity for my low usage personal DID |
| 13:10.07 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 13:10.31 | ManxPower | After Katrina they were VERY helpful to us |
| 13:11.21 | *** part/#asterisk screenn (~screenn@users-nat.more.com.ua) |
| 13:13.43 | Katty | so. |
| 13:13.58 | Katty | if anyone is interested in participating in the Asterisk Christmas Card Exchange, please /query me (= |
| 13:14.30 | ManxPower | <-- devout atheist |
| 13:14.43 | Kobaz | haha |
| 13:14.58 | Kobaz | i'm an athiest too, but we do christmas |
| 13:15.17 | hrhrhr | what exactly is a christmas card exchange |
| 13:15.24 | Katty | ManxPower: so am i, what's that got to do with anything :P |
| 13:15.36 | Katty | hrhrhr: it's a list of names and addresses |
| 13:15.45 | Katty | hrhrhr: and while you don't have to send to everyone on the list, you can |
| 13:15.52 | Katty | hrhrhr: in turn, you get christmas cards. |
| 13:15.59 | Kobaz | with money in them? |
| 13:16.06 | Katty | pfft |
| 13:16.17 | Katty | hand written note! |
| 13:16.36 | ManxPower | "I don't know who you are and I don't care. happy xmas!" |
| 13:16.49 | hrhrhr | bah humbug etc |
| 13:16.51 | hrhrhr | :P |
| 13:16.51 | Katty | hence the part about not sending to everyone ;) |
| 13:17.10 | Katty | it's ok manx, you don't have to participate |
| 13:17.21 | Katty | i'm not going to twist your arm about it |
| 13:17.32 | ManxPower | Kobaz, human history has a long tradition of a winter festival. But I still don't celebrate Christmas. |
| 13:19.28 | Kobaz | well, i don't really celebrate the way most people would |
| 13:20.02 | Kobaz | my family likes to get each other stuff, and we have some sort of dinner |
| 13:20.04 | Kobaz | that's about it |
| 13:21.36 | Katty | that sounds nice. |
| 13:21.53 | Katty | my family doesn't celebrate, but i get together with friends and have dinner. and then drinks. |
| 13:21.54 | Kobaz | yeah, it's an excuse to see each other |
| 13:22.17 | Katty | (= |
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| 13:25.23 | ariel_ | Morning, folks |
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| 13:36.48 | bip | I have a TDM410 card connected to the wall plug but in cannot dial into it, it has 3 FXO modules and one is plugged, I m using the latest asteriskNow 32 bit |
| 13:37.51 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 13:43.25 | Kobaz | bip: #asterisknow |
| 13:43.53 | bip | grazie Kobaz |
| 13:43.56 | bip | oops |
| 13:44.00 | bip | thanx Kobaz |
| 13:44.17 | bip | didn t know they had their own chan :) |
| 13:47.26 | Kobaz | yeah asterisknow is a gui on top of asterisk... it's hard to debug someone's asterisknow for us... we just work with raw asterisk |
| 13:48.09 | p3nguin | Actually, AsteriskNOW is a complete distribution. FreePBX or Asterisk GUI would be the GUI. |
| 13:49.01 | bip | well in #frepbx they asked me about thhe asterisk command line ... |
| 13:49.26 | Kobaz | well, yeah asterisknow is a bunch of stuff |
| 13:49.26 | bip | and dadhi commands |
| 13:50.00 | bip | but they r silent now so i decided to whine a tad here ;-) |
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| 14:01.36 | *** join/#asterisk mmlj4 (~jkelly@ip70-171-94-246.no.no.cox.net) |
| 14:07.09 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 14:10.45 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 14:11.01 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 14:13.27 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:16.56 | *** join/#asterisk b0gatyr (~b0gatyr@host-208-88-126-198.biznesshosting.net) |
| 14:17.10 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 14:24.09 | goddva | if i checkout a branch, from lets say olle, how could I do a simple diff with a branch that was released (1.4.36) to get a diff file(s)? |
| 14:25.16 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:25.28 | goddva | I probably dont want all the new changes merged from trunk... |
| 14:26.29 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 14:29.58 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:31.18 | kaldemar | goddva: diff -r -u path/to/release path/to/newbranch |
| 14:34.24 | goddva | kaldemar: but if the branch is in automerge, that would mean that the newbranch has a lot of other new code as well... right? |
| 14:35.02 | goddva | kaldemar: should i then checkout the "newbranch" with the same rev as the releaase? and then do the diff? |
| 14:51.32 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:51.51 | ManxPower | goddva, #asterisk-dev might be a good place to ask |
| 14:53.18 | goddva | ManxPower: true.. :) thanx |
| 14:55.18 | *** join/#asterisk timnis (~timnis@ha98.netikka.fi) |
| 14:57.28 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 15:01.00 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 15:03.24 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:05.20 | Katty | if anyone is interested in participating in the Asterisk Christmas Card Exchange, please let me know! |
| 15:06.29 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 15:07.14 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 15:08.15 | *** join/#asterisk timnis (~timnis@YZMMCCCXXXIV.gprs.sl-laajakaista.fi) |
| 15:11.47 | *** join/#asterisk cjk (~cjk@85.93.217.128) |
| 15:13.02 | cjk | hi, my fxs ports stop working after a few days (sometimes 2 days, sometimes 30) the only way to get them working again is unloading the kernel modules and reloading them. any ideas or hints |
| 15:14.10 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:14.49 | SaiSoma | Looking for a suggestion on updating voicemail.conf . ..I'm adding a section for asterisk into our existing web-based management system for *. It will create entries in sip.conf and extensions.conf, no problem, but I'm wondering about vm. |
| 15:15.58 | SaiSoma | should I change voicemail.conf to a sql config? if so, does anyone know where updated instructions on that might be? I find lots of older setups, but not certain if they will work in 1.8.0 |
| 15:36.52 | phix | :D |
| 15:39.06 | bneff | ~instantmessage |
| 15:41.29 | *** join/#asterisk fauxalliance (~gerald@207.231.237.59) |
| 15:43.45 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 15:45.35 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 15:47.13 | Katty | anyone have any thoughts on this? http://pastebin.ca/1982953 i'm having dahdi problems. |
| 15:48.09 | Qwell | Katty: it's already loaded. unload it first |
| 15:49.03 | Katty | Qwell: http://pastebin.ca/1982955 |
| 15:49.19 | Katty | Qwell: or did you mean, unload it, restart it, load it |
| 15:49.59 | *** join/#asterisk PotatoHead (~chatzilla@137.140.108.213) |
| 15:51.39 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 15:52.09 | p3nguin | What's a typical number of bits for an ssh rsa key on a router? The default is 512 if I don't enter any other amount. Is 768 good enough? Should I use 1024? |
| 15:53.10 | Katty | No hardware timing source found in /proc/dahdi, loading dahdi_dummy <- is that normal? |
| 15:53.30 | chuckf | p3nguin: if you're using anything modern there's no reason to use a key less than 1024 |
| 15:54.52 | p3nguin | chuckf: It allows me to choose between 360 and 2048, but I knew I wanted more than the default of 512. |
| 15:55.44 | chuckf | p3nguin: for the systems that I use ssh keys on most of them are 2048 just because the overhead doesn't hurt me at all |
| 15:58.33 | fullstop | cpu cycles are cheap these days. |
| 15:58.35 | fullstop | go 2048 |
| 15:59.10 | fullstop | and, honestly, how much time will you be spending in the router with an ssh console after things are configured? |
| 16:00.58 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 16:04.36 | Katty | anyone have any thoughts on what might be causing this dahdi error? http://pastebin.ca/1982971 |
| 16:05.59 | *** join/#asterisk AastraUser (~chatzilla@137.140.108.213) |
| 16:11.27 | *** join/#asterisk fofware (~Fabian@host184.190-226-209.telecom.net.ar) |
| 16:12.58 | *** join/#asterisk sbszulu (~dundubala@41.16.83.129) |
| 16:15.55 | p3nguin | It takes a long time to generate 2048 bit RSA keys on this system. |
| 16:16.30 | AastraUser | Has anyone here worked with an Aastra (formerly Ericsson) Dialog 422 IP phone in Asterisk before? I'm very new to Asterisk and have the basic "phone" functionality working, but I'd like to be able to enable some features like a second line, transfer, etc, and I was hoping to find a profile or guide that could help me out. |
| 16:18.59 | krion | hey guys i'm missing vm-tmpexists in en and fr |
| 16:19.06 | krion | anyone with a link in order to get it ? |
| 16:21.25 | krion | i found it, i guess :) |
| 16:21.39 | thehar | skips |
| 16:23.55 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 16:23.59 | Katty | well. |
| 16:24.01 | Katty | it was a bad card. |
| 16:24.11 | Katty | first bad sangoma card i've ever seen, too |
| 16:24.27 | Kobaz | wiggity |
| 16:24.41 | Katty | it was weird. and spastic. |
| 16:26.24 | thehar | oh was it |
| 16:26.44 | Katty | has another one in there now |
| 16:27.01 | mmlj4 | I loathe sangoma |
| 16:28.02 | mmlj4 | AastraUser: aastra phones are phones, they work. What you want is to make asterisk work. |
| 16:28.04 | Katty | i've had good luck with them, actually |
| 16:28.07 | fullstop | dumb question.. how do I Goto current priority + N? |
| 16:28.09 | Katty | anyone where work for sangoma? |
| 16:28.19 | mmlj4 | AastraUser: the logic is in asterisk |
| 16:28.52 | *** join/#asterisk BSarandan (~BSarandan@dsl092-128-191.chi1.dsl.speakeasy.net) |
| 16:30.11 | BSarandan | Hi all , Im tryin gto install Aterisk 1.8 and I keep getting this error: |
| 16:30.13 | BSarandan | collect2: ld returned 1 exit status |
| 16:30.13 | BSarandan | make[1]: *** [asterisk] Error 1 |
| 16:30.13 | BSarandan | make: *** [main] Error 2 |
| 16:30.20 | BSarandan | Can anyone please help me ? |
| 16:30.24 | ManxPower | I love Sangoma |
| 16:30.31 | AastraUser | mmlj4: I was thinking t was something in the configuration file on the phone I needed to edit. These phones are all Ericsson branded since they were made before the buyout. My main problem is that I can't get the phone to answer line 2 when it rings. I can get both lines to ring, but the phone will only allow me to answer one at a time. |
| 16:30.45 | ManxPower | BSarandan, how much memory does the system have |
| 16:30.45 | mmlj4 | are they SIP phones? |
| 16:30.56 | BSarandan | 4gb |
| 16:31.01 | ManxPower | Katty, Sangoma has official support on #sangoma |
| 16:31.11 | ManxPower | BSarandan, no VM or anything? |
| 16:31.14 | BSarandan | brand new supermicro server |
| 16:31.15 | BSarandan | no vm |
| 16:31.19 | BSarandan | just centos |
| 16:31.21 | ManxPower | BSarandan, and there is no such thing as 1.8 asterisk. |
| 16:31.28 | mmlj4 | well, they may have some setting, or require firmware... I know nothing about those phones, except that I liked the feel of them in my hand at astricon |
| 16:31.30 | BSarandan | and then asterisk |
| 16:31.40 | Katty | ManxPower: oooh |
| 16:31.46 | mmlj4 | as opposed to horrible polycom |
| 16:31.56 | Katty | ManxPower: fancy. |
| 16:32.03 | BSarandan | there is Asterisk 1.8.0 ... just downlaoded from the website |
| 16:32.12 | ManxPower | Katty, see the topic to see how to get support's attention |
| 16:32.27 | ManxPower | BSarandan, Someone said 1.8.0.1 is out |
| 16:32.52 | BSarandan | Let me see |
| 16:32.53 | Qwell | someone is wrong |
| 16:32.58 | fullstop | for inexpensive phones which work reasonably well, polycom works fine for me. |
| 16:33.00 | p3nguin | Not according to asterisk.org. |
| 16:33.09 | fullstop | oh snap |
| 16:33.09 | BSarandan | I`ve looked yesterday and there was only 1.8.0 |
| 16:33.22 | p3nguin | http://www.asterisk.org/downloads <-- no 1.8.0.1 here |
| 16:33.24 | Qwell | BSarandan: pastebin the entire output of make |
| 16:33.25 | Qwell | ~pb |
| 16:33.25 | 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.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. |
| 16:33.31 | BSarandan | sure |
| 16:33.33 | ManxPower | BSarandan, look thru all the compile output for errors, chances are the real error happened much earlier |
| 16:33.57 | BSarandan | yes, something like: |
| 16:33.58 | BSarandan | configure:6535: checking if the linker (/usr/bin/ld) is GNU ld |
| 16:34.01 | ManxPower | I'll wait for 1.8.20 thankyouverymuch 8-) |
| 16:34.15 | BSarandan | "/usr/bin/ld: cannot find -lpthreads |
| 16:34.20 | ManxPower | there you go |
| 16:34.41 | ManxPower | shouldn't ./configure have barfed if it didn't find libpthreads |
| 16:34.50 | Qwell | No, because that isn't an error. |
| 16:34.53 | BSarandan | that I have in config.log file |
| 16:35.00 | BSarandan | ./configure went well |
| 16:35.04 | BSarandan | make menuconfig ok again |
| 16:35.07 | ManxPower | Qwell, Asterisk can build without pthreads? |
| 16:35.10 | BSarandan | and when I hit make I get that error |
| 16:35.11 | Qwell | which is why I asked for the output of make. |
| 16:35.35 | ManxPower | BSarandan, less talk, more pastebin |
| 16:35.57 | BSarandan | this is the output of make: |
| 16:35.59 | BSarandan | chan_mgcp.c:(.text+0xaf04): undefined reference to `ast_pktccops_gate_alloc' |
| 16:35.59 | BSarandan | ../channels/chan_mgcp.eo: In function `start_rtp': |
| 16:35.59 | BSarandan | chan_mgcp.c:(.text+0xbdb2): undefined reference to `ast_pktccops_gate_alloc' |
| 16:35.59 | BSarandan | collect2: ld returned 1 exit status |
| 16:35.59 | BSarandan | make[1]: *** [asterisk] Error 1 |
| 16:35.59 | BSarandan | make: *** [main] Error 2 |
| 16:36.10 | ManxPower | BSarandan, INSTEAD OF FLOODING THE CHANNEL USE PASTEBIN |
| 16:36.18 | BSarandan | ok, sorry |
| 16:41.07 | russellb | BSarandan: you need res_pktccops |
| 16:41.22 | russellb | what, output of make? that's odd. o.O |
| 16:41.26 | russellb | s/what/wait/ |
| 16:41.43 | Qwell | russellb: 4.1, s/use/depend/;s/depend/makefunky/ |
| 16:41.52 | russellb | Qwell: ugh |
| 16:42.16 | russellb | BSarandan: try to upgrade gcc. :-) |
| 16:42.17 | Qwell | still waiting for the entire output of make, heh |
| 16:42.27 | russellb | l u n c h |
| 16:42.59 | BSarandan | gcc is version gcc-4.1.2-48.el5 |
| 16:48.29 | BSarandan | I`ve tried now with res_pktccops and it didn't work |
| 16:48.53 | timnis | Hi, have somebody get SIP TLS working with 1.8.0? I can register but when try to call I get ": FILE * open failed!"? |
| 16:50.25 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 16:52.57 | kjs | hm, if i want an phone to display "DID" when someone is rining my direct dial what do I need to look up in the manual? |
| 16:54.19 | Katty | if anyone is interested in joining the asterisk christmas card exchange, please /query me! |
| 16:55.32 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:55.32 | beek | Katty: How about a Solstice card exchange? |
| 16:55.37 | Katty | that would work. |
| 16:55.42 | Katty | also Yule |
| 16:55.56 | beek | Saves the annoyance of the various religions/nonreligions... |
| 16:56.09 | Katty | (= |
| 16:57.29 | Katty | you should join the list beek |
| 16:57.39 | Katty | and Qwell should too! |
| 16:57.49 | paulc | kjs: Set(CALLERID(name)=DID:${CALLERID(name)}) will do the trick |
| 16:57.55 | beek | Katty: How many are on the list right now? |
| 16:58.04 | Katty | checks |
| 16:58.14 | paulc | kjs: I use something similar to show me which DID was called, rather than using separate line buttons etc |
| 16:58.34 | Katty | beek: 5 |
| 16:59.09 | beek | Do we program our Asterisk systems to call each and have a recording by Allison that is played? |
| 16:59.11 | beek | :D |
| 16:59.16 | Katty | >.< |
| 16:59.32 | Katty | no you send a shiny card in the mail with a hand written note you goofball! |
| 16:59.50 | Katty | you don't have to send to everyone if you don't want to. |
| 17:00.05 | beek | That sounds like work. |
| 17:00.16 | Qwell | mails Katty to Zimbabwe |
| 17:00.23 | Katty | :< |
| 17:00.28 | Qwell | post-paid! |
| 17:00.32 | Katty | :<< |
| 17:00.48 | Katty | Qwell: you don't love me anymore :< there are germs there :<<< |
| 17:01.01 | Qwell | there are germs everywhere! |
| 17:01.13 | Qwell | would madagascar be better? they have beaches |
| 17:01.54 | Katty | how about portland oregon? |
| 17:02.04 | [TK]D-Fender | sends Katty to Abu-Dhabi |
| 17:02.07 | [TK]D-Fender | Get it right people! |
| 17:02.08 | Qwell | umm, I could do that |
| 17:02.08 | beek | Do that during OSCON |
| 17:02.11 | carrar | PDX!! |
| 17:02.14 | beek | (Portland) |
| 17:03.37 | Katty | zimbabwe has hepatitis a, hepatitis e, typhoid fever, malaria, dengue fever, yellow fever, japanese encephalitis, african trypanosomiasi, and many many other things :<<< |
| 17:04.38 | Qwell | Katty: Portland has rain! |
| 17:05.03 | Katty | i can live with rain |
| 17:05.17 | Katty | and my safe familiar diseases. |
| 17:05.18 | beek | Portland is the friendliest place I've ever visited. |
| 17:05.22 | Katty | like my safe familiar american BS |
| 17:05.32 | Qwell | no fun |
| 17:05.47 | Katty | no, no fun. |
| 17:06.10 | Katty | but happily sterile! |
| 17:13.32 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 17:15.20 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 17:28.34 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 17:31.18 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 17:33.43 | fullstop | madigascar also has lemurs. They are awesome. |
| 17:34.05 | coppice | fullstop: we all saw the movie |
| 17:34.38 | fullstop | The movie wasn't that great. |
| 17:34.54 | fullstop | Lemurs, however, are awesome when you get to watch them leap from tree to tree. |
| 17:35.16 | fullstop | Not as cute as tarsiers, but they are fun little animals. |
| 17:39.10 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 17:41.09 | *** join/#asterisk x86 (~x86@i.am.leet.org) |
| 17:41.17 | x86 | [TK]D-Fender: hey man ltns... |
| 17:42.00 | [TK]D-Fender | y0 |
| 17:42.03 | x86 | I've got a Polycom IP601 that I can not get to register with my new 1.8.0 system no matter what I do... sip debug keeps telling me it's returning a 401 unauthorized code when the phone tries the register |
| 17:42.23 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 17:42.31 | x86 | I've got the phone setup to register to the IP of the asterisk box, no hostnames used anywhere |
| 17:42.56 | x86 | some website mentioned I should add domain=10.10.10.4 to sip.conf (where 10.10.10.4 is * server) |
| 17:43.33 | x86 | but that did not help... I also verified the username (7899, same as exten), and password (matches exten also) were the same on both sides.... |
| 17:43.44 | x86 | I think it's a domain and/or realm issue... any ideas? |
| 17:44.17 | [TK]D-Fender | ~pb |
| 17:44.17 | 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.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. |
| 17:44.25 | x86 | I'm not specifing any "realms" at all... |
| 17:44.30 | x86 | [TK]D-Fender: you want my sip.conf? |
| 17:44.46 | [TK]D-Fender | x86: configs, SIP debug, etc |
| 17:45.36 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 17:46.37 | x86 | [TK]D-Fender: http://pastebin.ca/1983065 |
| 17:46.48 | x86 | that's the sip.conf in entirety |
| 17:46.59 | Katty | if anyone is interested in joining the asterisk christmas card exchange, please /query me! |
| 17:47.57 | x86 | [TK]D-Fender: http://pastebin.ca/1983066 |
| 17:48.02 | x86 | heya Katty! ltns |
| 17:48.10 | x86 | no interest in xmas cards tho :P |
| 17:49.06 | [TK]D-Fender | x86: Polyc oncifgs now... |
| 17:49.15 | Katty | x86: k (= |
| 17:49.17 | Katty | hugs x86 |
| 17:49.19 | x86 | it's all config'd by the menu on the phone for now |
| 17:49.50 | x86 | [TK]D-Fender: I'm not yet provisioning it... I lost all my old templates and sip.ld's and such :( |
| 17:50.11 | x86 | [TK]D-Fender: perhaps you have a stash you could tar up and send me? me love you long time! :) |
| 17:50.44 | [TK]D-Fender | Nothing says "I love you" like PayPal :p |
| 17:51.26 | x86 | nothing says "i love you" like helping someone out who speaks to you as a vietnamese prostitute either ;-) |
| 17:52.07 | x86 | do you see anything wrong with the sip.conf or anything? |
| 17:52.21 | x86 | any oddities with the sip debug? I couldn't identify anything |
| 17:52.40 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 17:53.23 | *** join/#asterisk atan2 (~atan@blk-222-132-103.eastlink.ca) |
| 17:53.52 | x86 | fwiw, this is my home single-phone system heh |
| 17:54.17 | x86 | not like I'm getting billable hours for your help :P |
| 17:54.41 | *** join/#asterisk coppice (~chatzilla@116.92.195.24) |
| 18:05.04 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 18:05.35 | pabelanger | x86: any specific reason for setting compactheaders = yes ? |
| 18:06.02 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 18:22.12 | x86 | pabelanger: no, you think it would help to eliminate that? |
| 18:22.27 | *** join/#asterisk grabes (~grabes@70.15.27.211.res-cmts.sm.ptd.net) |
| 18:23.03 | grabes | DId asterisk start some new licensing for g.729 on asterisk 1.6.2.13? I use freebsd jails as a test environment, and now my coec for g.729 will not load |
| 18:23.04 | x86 | didn't help |
| 18:23.10 | pabelanger | x86: No, I've never seen anybody use it. I was curious if you have a reason too. |
| 18:23.20 | x86 | ah |
| 18:23.30 | x86 | well.... I'm stumped... any ideas would be awesome |
| 18:23.54 | Qwell | grabes: Did you upgrade the module? |
| 18:24.13 | *** join/#asterisk Tim_Toady (~moi@77.49.109.71) |
| 18:24.19 | grabes | I recompiled it yes, its the free one that I use for testing. |
| 18:24.24 | Qwell | wait, no. There is no FreeBSD version for 1.6+ |
| 18:24.48 | pabelanger | ~collectdebug |
| 18:24.48 | infobot | i heard collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 18:24.55 | pabelanger | x86: grab a full debug and PB it |
| 18:25.13 | pabelanger | I'm curious if you having the same issue I found this morning |
| 18:25.23 | pabelanger | I don't think so however |
| 18:26.02 | grabes | Qwell: I use the opensource g.729 for freebsd |
| 18:26.29 | Qwell | rolls his eyes |
| 18:26.39 | fauxalliance | pukes a little |
| 18:26.51 | Qwell | it's not open source. no more discussion of it here. |
| 18:27.05 | fauxalliance | take it back to Latvia. |
| 18:27.15 | grabes | Qwell: whatever, do you want to see my fucking licenses for my linux gateways. Sorry I use FreeBSD for quick testing becuase of its jails |
| 18:27.17 | fauxalliance | misses Riga |
| 18:27.33 | pabelanger | <PROTECTED> |
| 18:27.45 | Qwell | grabes: Licenses are done on an implementation basis. It's not ME that gives a shit. It's the patent-holders and the copyright-holders. |
| 18:27.49 | fauxalliance | revokes grabes 'fucking license' |
| 18:28.23 | Qwell | You can discuss it with them. You aren't getting help with it here. |
| 18:28.38 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 18:28.43 | fauxalliance | ^^ moot point |
| 18:28.55 | Qwell | If you have an issue with your licensed copy, great - call Digium support and they'll be happy to help you. |
| 18:29.34 | p3nguin | Hmm. Is there a free or open source g.729 for _ANY_ platform? |
| 18:29.36 | grabes | rolls eyes, and pukes a little |
| 18:29.39 | Qwell | p3nguin: no |
| 18:30.03 | p3nguin | So where did this misconception come from? |
| 18:30.09 | Qwell | and even if there were, it wouldn't be legal to use. |
| 18:30.17 | Qwell | because people are dumb? |
| 18:30.20 | jdoe | p3nguin: the intel thing, presumably. |
| 18:30.26 | grabes | Qwell: I hope one day you j-walk... |
| 18:30.31 | Qwell | jdoe: No, that violates *INTEL*s license. |
| 18:30.50 | Qwell | grabes: My friend once got a ticket for gang-related jay-walking. |
| 18:30.51 | jdoe | Qwell: I should have clarified... what I mean is most people associated free-to-download with free-to-use |
| 18:30.56 | jdoe | er, associate |
| 18:31.00 | x86 | pabelanger: see the above pastebin, already done |
| 18:31.02 | Qwell | jdoe: It's neither. |
| 18:31.02 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 18:31.22 | p3nguin | He mentioned something about compiling a module... |
| 18:31.48 | Qwell | Using illegally obtained software. :) |
| 18:32.19 | Qwell | gee, I wonder why it's hosted from Latvia. *rolleyes* |
| 18:32.22 | p3nguin | Oh, I see. I didn't know the source was available. |
| 18:32.38 | grabes | Qwell: where can I purchase the 1 license I need for my FreeBSD test box? |
| 18:32.40 | jdoe | Qwell: hrm. Download page (Intel's) has changed, I could've sworn it was freely available before. Shrug. I dunno man, I don't work here. |
| 18:32.46 | grabes | Qwell: I need to unravel your panties |
| 18:32.53 | Qwell | grabes: There are no FreeBSD versions at this time. |
| 18:33.00 | x86 | Qwell: perhaps you can tell me why my Polycom IP601 phone gets a 401 when it tries to register to * 1.8.0? |
| 18:33.18 | jdoe | x86: credential problem? |
| 18:33.24 | pabelanger | x86: debug output too, not just sip |
| 18:33.25 | Qwell | x86: There was a post on the mailing list, and now a bug on mantis.. something about a tab in a nonce. Kinda weird. |
| 18:33.37 | Qwell | though unlikely, it's possible that you're hitting the same issue. |
| 18:33.42 | x86 | http://pastebin.ca/1983065 <-- sip.conf |
| 18:33.59 | x86 | http://pastebin.ca/1983066 <-- sip debug |
| 18:34.20 | fauxalliance | p3nguin, just some sudo educational licence from IBM... but technically, no. no freebies |
| 18:34.21 | Qwell | nope, no nonce tab |
| 18:34.27 | Qwell | fauxalliance: Intel* |
| 18:34.31 | pabelanger | http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt <-- what we need |
| 18:34.44 | fauxalliance | Qwell, better the devil _you_ know ;-P |
| 18:35.31 | *** part/#asterisk Khratos (~jespinal@66.128.60.148) |
| 18:36.02 | Qwell | grabes: It seems rather silly to have a test system be a different OS than your live system.. |
| 18:36.18 | fauxalliance | ...for education purposes only and if a patent exists in your country for G.729 or G.723.1 then you should contact the owner of that patent and request their permission before executing the code..... |
| 18:36.28 | *** join/#asterisk devmod (~devmod__@c-76-100-208-204.hsd1.md.comcast.net) |
| 18:36.40 | Qwell | fauxalliance: and even then - you have to license the code from Intel. Nobody else has the right to distribute that. |
| 18:36.49 | fauxalliance | perhaps the info bot should know about that... |
| 18:37.01 | Qwell | and EVEN THEN, there are about 2 uses for it. no other use of it is legal. |
| 18:37.23 | Qwell | testing != education |
| 18:37.26 | grabes | Qwell: I don't think Intel has anything to do with the g.729 |
| 18:37.43 | fauxalliance | ^^ is mistaken |
| 18:37.53 | grabes | Qwell: Unless you are referring to IPP? |
| 18:38.02 | Qwell | It's their copyright you're violating when you use the illegal one. |
| 18:38.43 | grabes | http://en.wikipedia.org/wiki/G.729 |
| 18:38.50 | fauxalliance | grabes, consider the source |
| 18:39.17 | Qwell | this discussion is done. |
| 18:39.23 | fauxalliance | G.729 includes patents from several companies and is licensed by Sipro Lab Telecom,,, guess who is one the several companies grabes |
| 18:39.35 | jdoe | grabes: the "opensource" g729 module wraps IPP. |
| 18:39.49 | fauxalliance | which also requires licensing :P |
| 18:39.59 | Qwell | drop it. |
| 18:40.09 | grabes | Qwell: I didn't realize you were an admin |
| 18:40.12 | grabes | please end it |
| 18:40.13 | fauxalliance | drops in favour of Dr. Pepper. |
| 18:40.27 | jdoe | lol |
| 18:40.29 | Qwell | fauxalliance: fwiw, Digium accepts pallets of Dr. Pepper. |
| 18:40.38 | fauxalliance | compliments Qwell on his hat... |
| 18:40.48 | Qwell | fauxalliance: were you at astricon? |
| 18:40.57 | fauxalliance | Qwell, no sir. i wish i could go |
| 18:41.03 | Qwell | oh. different hat then. :p |
| 18:41.10 | fauxalliance | I was at the solid waste convention in Texas |
| 18:41.12 | Qwell | we had some pretty sweet hats at the all-conference party |
| 18:41.35 | fauxalliance | Qwell, the 'operator' hat ;-)... have a pic of the other... i am intrigued now |
| 18:41.39 | Qwell | we all woke up with glitter *all over us* |
| 18:41.41 | Qwell | good times. |
| 18:41.55 | fauxalliance | Qwell, all in the name of telephony ! |
| 18:42.01 | Qwell | indeed |
| 18:42.22 | fauxalliance | I will hit astricon one of these days... but mostly... I have to play with Garbage.... |
| 18:45.43 | [TK]D-Fender | plays Garbage... |
| 18:45.48 | [TK]D-Fender | at least ONE song... |
| 18:45.49 | fauxalliance | Ms. Manson? |
| 18:45.54 | [TK]D-Fender | Indeed |
| 18:46.00 | fauxalliance | asks Katty to dance |
| 18:46.42 | fauxalliance | [TK]D-Fender, what one? |
| 18:47.03 | [TK]D-Fender | fauxalliance: "Stupid Girl" |
| 18:47.13 | [TK]D-Fender | fauxalliance: I'm on keys for that one |
| 18:47.33 | fauxalliance | cool... i have a cool remix of 'only happy when it rains' |
| 18:47.39 | fauxalliance | ^^my favourite |
| 18:48.19 | Qwell | so uhh |
| 18:48.23 | Qwell | yeah. |
| 18:48.29 | Qwell | we're all old. that was FIFTEEN YEARS AGO. |
| 18:48.36 | fauxalliance | [TK]D-Fender, have any of the old angelfish work? |
| 18:49.07 | [TK]D-Fender | ? |
| 18:49.16 | Katty | dances with fauxalliance! |
| 18:49.18 | fauxalliance | [TK]D-Fender, her _original_ band |
| 18:49.33 | petern_ | i have some |
| 18:49.35 | fauxalliance | and Katty dances! |
| 18:49.37 | *** join/#asterisk knot (~knotsucke@unaffiliated/devemo) |
| 18:49.40 | petern_ | it's depressing ;p |
| 18:50.05 | fauxalliance | petern_, thats why i like it so much. |
| 18:50.23 | [TK]D-Fender | fauxalliance: Never heard of... |
| 18:51.08 | [TK]D-Fender | Most of my music is 20-25 years old... |
| 18:51.29 | fauxalliance | [TK]D-Fender, modern music is too middle of the road, too much pitch shifting, not enough passion |
| 18:52.01 | fauxalliance | [TK]D-Fender, leah andreone? |
| 18:53.15 | [TK]D-Fender | Never heard of... |
| 18:53.51 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:54.17 | fauxalliance | [TK]D-Fender, Melanie Fiona? http://www.youtube.com/watch?v=H8yCLErPeP8&feature=related she is new |
| 18:55.46 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 18:57.20 | fauxalliance | was just busted... Groovin' in a posted NO-GROONIN' zone... |
| 18:57.32 | *** join/#asterisk BMJ (~bjohns@c-24-126-158-110.hsd1.ga.comcast.net) |
| 18:57.32 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 18:57.33 | [TK]D-Fender | fauxalliance: Heard the name, sure I've heard at least one song, but couldn't tell you it by name.. |
| 18:58.30 | *** join/#asterisk [T]ank (~chwall@206.71.78.158) |
| 18:58.45 | [TK]D-Fender | fauxalliance: "Give It To Me Right". Thats the one |
| 18:58.49 | [TK]D-Fender | fauxalliance: Annoying :p |
| 19:00.53 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 19:01.51 | Katty | glomps jblack |
| 19:02.39 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 19:04.53 | *** join/#asterisk diemos (~diemos@173-13-138-49-sfba.hfc.comcastbusiness.net) |
| 19:09.53 | *** join/#asterisk wfsystems (~wfsystems@office.cmh.ewebforce.net) |
| 19:10.58 | wfsystems | our asterisk server is jacked. both our PRI lines are working, but asterisk is giving messages about lagged and unreachable clients. we're kind of bad at troubleshooting. any suggestions? |
| 19:12.35 | Qwell | Check why the endpoints are lagged/unreachable. |
| 19:12.44 | Qwell | (hint: it's likely your network/internet connection) |
| 19:17.47 | Kobaz | interwebs |
| 19:18.00 | Katty | ^- i like dem. |
| 19:18.30 | Qwell | I'd be a very different person without the interwebs. |
| 19:19.01 | Kobaz | it's so weird... thinking about before the interwebs... what did people do on computers? |
| 19:19.11 | Qwell | BBS |
| 19:19.13 | frigidzephyr | books |
| 19:19.15 | Kobaz | i mostly played games on my apple [] |
| 19:19.19 | Qwell | frigidzephyr: ...gtfo |
| 19:19.27 | frigidzephyr | sulks |
| 19:19.30 | Kobaz | i was on some bbs's |
| 19:19.42 | Qwell | books. pfft! :D |
| 19:19.45 | Kobaz | mostly it was just like collections of shareware |
| 19:19.55 | Qwell | Kobaz: and porn. |
| 19:20.02 | Kobaz | well yeah, of course |
| 19:20.22 | Katty | aww :< |
| 19:20.24 | Katty | hugs frigidzephyr |
| 19:20.31 | Katty | no sulking allowed on friday! |
| 19:20.35 | Katty | it's a rule, i just made up! |
| 19:20.38 | wfsystems | good call folks. i think we're being ddos'd or something |
| 19:20.40 | Kobaz | er... make that apple ][ |
| 19:20.43 | Katty | and i'm backing up by this ......... |
| 19:20.54 | Katty | not-gun. not lent to me by the national rifle association |
| 19:20.58 | Qwell | wfsystems: you are not the first to mention that in recent days. |
| 19:21.00 | *** join/#asterisk gregd (~gregd@188-220-38-34.zone11.bethere.co.uk) |
| 19:21.40 | Katty | i can't honestly recall what i did before the days of internets. |
| 19:21.43 | Qwell | the breadth of these recent attacks are...large. |
| 19:21.48 | Katty | but i think it was probably going to school. |
| 19:21.58 | wfsystems | that's bad news =/ |
| 19:22.02 | Kobaz | i was writing code for apple and cp/m |
| 19:22.17 | Katty | Kobaz: that sounds very nerdy. |
| 19:22.21 | Kobaz | yeah |
| 19:22.34 | Kobaz | mostly just simple stuff... i didn't start real programming until i was in junior high or so |
| 19:22.59 | Kobaz | Katty: cp/m was the grandfather of DOS |
| 19:23.01 | Katty | also nerdy. |
| 19:23.11 | Katty | did you chase girls in junior high? |
| 19:23.22 | Kobaz | good question |
| 19:23.25 | *** join/#asterisk mandragor (~ergudicsu@70.158.116.62) |
| 19:23.28 | Katty | oh |
| 19:23.33 | Katty | you were too busy programming to notice? |
| 19:23.41 | Kobaz | heh... i did go after a few |
| 19:23.45 | Katty | excellent. |
| 19:23.50 | Kobaz | they were mostly interested in high school guys who played football |
| 19:23.57 | Katty | i dated a football player |
| 19:23.59 | Katty | he was a jerk |
| 19:24.01 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:24.06 | Katty | HAI CHAINSAW! |
| 19:24.31 | Katty | Kobaz: the nerdy sorts are much much much better. |
| 19:24.52 | Chainsaw | Katty: Hello there! :) |
| 19:24.53 | Kobaz | i agre |
| 19:24.54 | Kobaz | e |
| 19:24.59 | fullstop | I read that as "nerdy sports" and was trying to figure out which sports were nerdy. |
| 19:25.03 | fullstop | Golf, I suppose |
| 19:25.07 | Chainsaw | Katty: How are you today? *hug* |
| 19:25.13 | Kobaz | aid climbing is nerdy |
| 19:25.35 | Katty | fullstop: hmm nerdy sports? hmm |
| 19:25.38 | Katty | hugs Chainsaw |
| 19:25.42 | Katty | Chainsaw: am goodly thanks :> |
| 19:25.50 | Katty | D&D could be a sport |
| 19:25.54 | fullstop | and I was thinking, "extreme" chess, no matter how "extreme" is not a sport. |
| 19:25.55 | Katty | if you nerdraged every roll |
| 19:26.01 | Katty | ULTIMATE FRISBY |
| 19:26.04 | Katty | that might be nerdy. kinda. |
| 19:26.07 | Katty | not really. |
| 19:26.10 | Kobaz | fullstop: chessboxing |
| 19:26.12 | Chainsaw | Extreme ironing comes to mind. |
| 19:26.18 | Katty | hehe |
| 19:26.31 | Chainsaw | I believe it has a wiki article and everything :) |
| 19:26.32 | fullstop | especially when you break out the starch |
| 19:26.43 | fullstop | or the "magic sizing" |
| 19:26.49 | *** join/#asterisk bmint (~brian@h247.166.117.75.dynamic.ip.windstream.net) |
| 19:27.12 | fullstop | clearly I know too much about ironing. |
| 19:27.17 | Katty | clearly. |
| 19:27.20 | Katty | i don't know /anything/ about ironing. |
| 19:27.33 | Kobaz | i have an extreme ironing calendar somewhere |
| 19:27.35 | Katty | of course nothing i wear needs to be ironed |
| 19:27.37 | *** join/#asterisk Khratos (~jespinal@66.128.60.148) |
| 19:28.11 | Kobaz | i think aid climbing is one of the most nerdy but physically exhausting sports |
| 19:28.15 | Katty | does anyone work with Twisted? |
| 19:28.31 | fullstop | python? |
| 19:28.32 | Kobaz | i mean who wouldn't want to take all this crap up a rock wall for a few days http://c0278592.cdn.cloudfiles.rackspacecloud.com/original/588546.JPG |
| 19:28.34 | fullstop | I've dabbled |
| 19:28.46 | Kobaz | so sexy |
| 19:28.57 | Katty | no i mean the person with the /nick twisted |
| 19:29.16 | Katty | better yet, does anyone here work for Asteria? |
| 19:29.27 | fullstop | This is far more fun: http://sphotos.ak.fbcdn.net/hphotos-ak-ash2/hs561.ash2/148323_577198372709_3805250_33857461_6904771_n.jpg |
| 19:29.29 | Kobaz | i met the owners of aAsteria at astricon |
| 19:29.48 | Katty | hasslehoff |
| 19:30.03 | wfsystems | Qwell: do you know more about these attacks? |
| 19:30.15 | Katty | Kobaz: hoffmeyer |
| 19:30.34 | Katty | Kobaz: james. |
| 19:30.38 | Qwell | wfsystems: nope, just whats been posted to the lists. It's been pretty heavy though... |
| 19:31.22 | wfsystems | i'll check the lists out. thanks =) |
| 19:33.23 | Kobaz | your name looks way too much like wtfsystems |
| 19:33.47 | *** part/#asterisk grabes (~grabes@70.15.27.211.res-cmts.sm.ptd.net) |
| 19:34.39 | citywok | well i found bugs in my exchange EWS implementation. having fixed them, * still segfaults. lmao |
| 19:35.42 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 19:35.53 | Chainsaw | citywok: You found bugs in Exchange. Wow. |
| 19:36.38 | wfsystems | thanks for the help guys! it seems to be under control for now. have a good weekend! |
| 19:37.00 | citywok | it wasn't a bug in exchange, it was a permissions setting (NTLM auth wasn't enabled). lol. |
| 19:37.29 | Kobaz | i need a chair with squishy arm rests |
| 19:37.56 | frigidzephyr | i do too |
| 19:37.59 | frigidzephyr | i want a motorized office chair |
| 19:42.00 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 19:42.25 | *** join/#asterisk ukine_droid (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 19:43.48 | lanning | Bring me the Comfy Chair! |
| 19:46.28 | Katty | puts a comfy chair on lanning's head |
| 19:46.46 | *** join/#asterisk JoseBravo (~Jose@190.144.124.194) |
| 19:46.48 | JoseBravo | Hello |
| 19:46.51 | Katty | ohai |
| 19:47.17 | JoseBravo | Can i check from my CLI what codecs are supported for audio in a remote SIP peer? |
| 19:47.56 | *** join/#asterisk boch (c829e3f5@gateway/web/freenode/ip.200.41.227.245) |
| 19:48.18 | boch | hello, does anybody has experience working with TDD? |
| 19:52.10 | gregd | guys, i've got a callgroup defined... however i'm not able to choose it as destination for an 'inbound routes', is there any reason for it? |
| 19:52.41 | gregd | sorry, i ment ringgroup |
| 19:53.01 | Katty | http://i.imgur.com/mIxUg.jpg <- and i thought /i/ was colorful |
| 19:53.23 | Katty | gregd: i think we should see some cli pastebin info |
| 19:53.59 | ManxPower | gregd, FreePBX/Trixbox is not supported here. |
| 19:54.19 | gregd | fair enough :) |
| 19:58.09 | Katty | Qwell: is it snowing out there too? |
| 19:58.22 | Qwell | I haven't seen any snow |
| 19:58.32 | Qwell | is twisted still in Huntsville? |
| 19:58.48 | Qwell | clouds definitely look snowish though |
| 19:58.55 | Katty | i think so |
| 19:59.36 | ManxPower | JoseBravo, no. you could turn on sip debug and see what list of codecs the phone sends to asterisk when trying to make a call |
| 19:59.43 | *** join/#asterisk jstapleton (~jstapleto@173-15-197-75-BusName-Richmond.hfc.comcastbusiness.net) |
| 20:00.12 | Katty | Qwell: according to his twitter checkins, yes he's in huntsville |
| 20:00.30 | Katty | Qwell: or was yesterday, at any rate |
| 20:01.30 | jstapleton | is it possible to set a hint for a call out to a certain number over a DAHDI group? For example, if someone calls my cell (DAHDI/g1/5551212), can I set a hint on that? |
| 20:02.01 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 20:04.06 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 20:04.28 | JoseBravo | ManxPower, Im getting the call from a Cisco Gateway, and I get this: Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) |
| 20:04.55 | JoseBravo | ManxPower, that means that the remote gateway only support g729 right? |
| 20:06.12 | ManxPower | JoseBravo, that means the gateway is CONFIGURED to only support G729 |
| 20:06.42 | ManxPower | unless you purchase a G729 license for Asterisk there is very little you can do with a G729 call |
| 20:07.11 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 20:07.45 | JoseBravo | ManxPower, I already have the g729 working fine. But I need to switch to g711 when a fax is detected. |
| 20:08.13 | ManxPower | JoseBravo, good luck with that. |
| 20:08.17 | drift- | how can i get NAT to say Y on my newtork , cable modem dmz to my linux router/asteirks machine , i'm trying to get my phone to work from outside internal network at home to connect to office asterisk and work, at moment i can recieve calls and make calls but have no volume :( |
| 20:08.39 | *** join/#asterisk mobileman (~dimmy@193.107.184.193) |
| 20:08.41 | drift- | nobody can hear me and i cant hear them |
| 20:08.42 | ManxPower | ~sipnat |
| 20:08.43 | infobot | somebody said sipnat was Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:08.51 | ManxPower | drift-, there are your nat DOCS |
| 20:09.05 | JoseBravo | ManxPower, hehehe too hard? |
| 20:10.24 | ManxPower | JoseBravo, google sip fax asterisk |
| 20:11.15 | JoseBravo | ManxPower, ok |
| 20:11.23 | mobileman | hello everybody! |
| 20:12.30 | JoseBravo | Thx |
| 20:13.15 | mobileman | can anybody help me with incoming sip fax detection and reinvite to g.711? is it possible? |
| 20:13.57 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 20:15.57 | Katty | weeeeeeeeeeeeee |
| 20:16.10 | Katty | infobot: whee |
| 20:16.11 | infobot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 20:16.43 | *** join/#asterisk simplydrew (~simplydre@66.181.225.250) |
| 20:18.30 | mobileman | er?... |
| 20:21.31 | bmint | I am having trouble registering our Asterisk server with our SIP provider. I can register with Xlite. Anyone have an example register string for Asterisk 1.6* |
| 20:23.26 | *** join/#asterisk n0tk (~n0tk@216.160.42.30) |
| 20:23.31 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:24.08 | gregd | anyone interested in helping me out analyzing my disconnect tone? i do not seem to find a proper pattern in it :/ http://img233.imageshack.us/i/27270791.jpg/ |
| 20:25.58 | Katty | what's that apt-get thing for distrobution update |
| 20:26.27 | wdoekes2 | apt-get dist-upgrade ? |
| 20:26.33 | Kobaz | apt-get dist-upgrade |
| 20:26.41 | *** join/#asterisk n3hxs (~HAMming@75-145-10-45-Delmarva.hfc.comcastbusiness.net) |
| 20:26.48 | Qwell | Ubuntu? update-manager -c |
| 20:26.50 | Katty | hot. thanks. |
| 20:27.35 | Qwell | wdoekes2: I do believe there's a way to get Asterisk to tell you what it sees. |
| 20:27.50 | Qwell | also you're doing it wrong. :D |
| 20:29.00 | *** join/#asterisk enterneo (~enterneo@firewallix.jacobs-university.de) |
| 20:29.11 | Qwell | wdoekes2: if it doesn't show anything useful, try setting dring1 et al, and set an absurdly large dring1range |
| 20:29.17 | Qwell | brute force it :p |
| 20:29.42 | Qwell | it *should* tell you what it actually detected. then you can set it from there |
| 20:29.49 | enterneo | I am using 50001 as bindport for IAX2, I am trying to register two asterisk servers using IAX (the other one is using default port), so I am able to register but the other server is not, any hints ? |
| 20:29.58 | Qwell | actually, -1 apparently will force a match always |
| 20:30.03 | wdoekes2 | Qwell: to whom are you talking? |
| 20:30.14 | Qwell | wdoekes2: You. Which would be why I prefixed my comment with your name. |
| 20:30.25 | Katty | grammar++ |
| 20:30.30 | wdoekes2 | I see that. but I have no idea why and what about |
| 20:30.41 | Qwell | d'oh. |
| 20:30.44 | Qwell | gregd: |
| 20:30.49 | wdoekes2 | :) |
| 20:30.52 | Qwell | wdoekes2: my bad. I don't know where I got your name from. |
| 20:31.20 | Katty | is there an easy way to mount a samba share from linux box B to linux box A |
| 20:31.52 | Katty | samba on b is already 100% functional |
| 20:32.02 | Qwell | Katty: usually `smbmount` will give you a good commandline to use to mount it |
| 20:32.14 | Katty | thank you dear. |
| 20:32.24 | Qwell | but it'd probably be something like `mount -t smbfs //server/share/ /local/path/to/mount/ |
| 20:33.00 | Katty | Qwell: smbfs? |
| 20:33.06 | Qwell | something like that |
| 20:33.10 | Katty | ah ha! |
| 20:33.12 | Qwell | smbfs or sambafs. I think it's smbfs though |
| 20:33.33 | Katty | we shall see. |
| 20:33.34 | Qwell | setting up the server side of things on Linux is a PITA.. not really familiar with that part. |
| 20:33.47 | Katty | i'm basically going to mirror two linux boxes. kinda. |
| 20:34.05 | Katty | not so much as mirror them, but cronjob a copy of the conf files between the two |
| 20:34.09 | Qwell | using nfs would be far better, if it's just Linux |
| 20:34.10 | wdoekes2 | yuck! |
| 20:34.15 | wdoekes2 | scp? |
| 20:34.23 | Katty | Qwell: oh? |
| 20:34.25 | Katty | Qwell: go on. |
| 20:34.34 | Katty | Qwell: i don't know much about file sharing between linux boxes. |
| 20:34.46 | Qwell | smb is a Windows thing. it sucks |
| 20:34.48 | wdoekes2 | Katty: password-less ssh key and scp |
| 20:35.15 | Katty | Qwell: well i have to have samba shares regardless |
| 20:35.27 | Katty | Qwell: which is why i kinda figured i'd use that |
| 20:35.33 | Katty | wdoekes2: would you be willing to show me how to do that? |
| 20:35.54 | Katty | wdoekes2: if not, i will do it my way |
| 20:36.03 | wdoekes2 | ssh-keygen[enter] copy ~/.id_rsa.pub to .ssh/authorized_keys on the other box, voila |
| 20:36.11 | Katty | mhmm. |
| 20:36.12 | Katty | yeah |
| 20:36.17 | Katty | goes back to doing it her way |
| 20:39.05 | Katty | wdoekes2: is /root/.ssh/authorized_keys a directory? |
| 20:39.16 | wdoekes2 | /root/.ssh is a dir |
| 20:39.21 | wdoekes2 | authorized_keys is a file |
| 20:39.27 | wdoekes2 | must be chmod 0600 |
| 20:39.33 | Katty | what is 0600 |
| 20:39.34 | Katty | and why |
| 20:39.55 | wdoekes2 | rw for user, nothing for group and nothing for world (other) |
| 20:39.56 | mobileman | so what about faxes?... |
| 20:40.03 | wdoekes2 | -rw------- |
| 20:40.26 | wdoekes2 | why, because ssh likes it to be kind of secret |
| 20:40.39 | Katty | i don't have an in-depth understanding of linux |
| 20:40.41 | Katty | i like asking why |
| 20:41.04 | n3hxs | Better to ask, than spend hours wondering Why! |
| 20:41.18 | n3hxs | Morning all! |
| 20:41.19 | Katty | n3hxs: it's good to know concepts. it helps prevent additional questions. |
| 20:41.24 | Katty | hugs n3hxs |
| 20:41.28 | n3hxs | Oh yes. |
| 20:41.39 | Katty | wdoekes2: thank you for the information. i am excited to try it out. |
| 20:42.53 | Katty | mobileman: i'm not a fan of using faxes/faxing with asterisk |
| 20:43.03 | Katty | mobileman: but what's your question |
| 20:44.04 | p3nguin | ssh-copy-id makes short work of copying keys from one system to the other. |
| 20:44.15 | mobileman | is it possible to detect incoming faxes via sip and then send reinvite to g.711 for transmission? |
| 20:46.35 | Katty | p3nguin: could you give me an example of how i would do that? |
| 20:46.35 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 20:46.42 | Katty | mobileman: not sure. never tried doing that before. |
| 20:46.59 | Katty | mobileman: something tells me a dumb fax just isn't going to get it tho. |
| 20:47.13 | Katty | mobileman: due to timing, and delay |
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| 20:50.37 | p3nguin | ssh-copy-id user@host |
| 20:54.33 | Katty | does that transfer the file to the appropriate directory? |
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| 20:54.59 | Kobaz | Katty: yes |
| 20:55.04 | Kobaz | it does everything for you |
| 20:55.15 | p3nguin | That's the whole point of the command. |
| 20:55.24 | Katty | what if you have multipe keys tho? |
| 20:55.29 | Katty | how does it know which key to copy to which box? |
| 20:55.34 | p3nguin | You'll only have one per user, right? |
| 20:55.39 | Katty | i have no idea. |
| 20:55.41 | Katty | i'm not familiar with keys |
| 20:55.49 | p3nguin | You'll only have one per user per host. |
| 20:55.55 | Kobaz | if you have multiple keys use the -i option to specify which one |
| 20:56.06 | Katty | per...user? |
| 20:56.13 | Katty | the one that i just generated is for... root then? |
| 20:56.13 | Kobaz | ssh-copy-id -i ~/.ssh/id_rsa.pub user@host |
| 20:56.22 | p3nguin | You might have both rsa and dsa keys, but one rsa is enough. |
| 20:56.36 | Kobaz | i dunno what you generated |
| 20:56.36 | Katty | we should talk about this monday |
| 20:56.49 | Katty | they're locking the building up |
| 20:56.51 | Kobaz | you just copy the key you use for the user that you want to log in from |
| 20:57.04 | Kobaz | use your katty-power to keep them away |
| 20:57.07 | n3hxs | See ya.. Off to the new house. :) |
| 20:57.39 | Kobaz | mmm, 2 million rows of queue log data... yay |
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| 21:56.16 | delroy | anyone ever used a Sangoma A400? |
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| 22:02.56 | JoseBravo | The codec of a sip channel can be changed on the fly? for example I receive a call using the g729 codec but when I send it to a specific extension can I change the codec to g711? |
| 22:16.28 | delroy | Anyone used Sagnoma TDM cards before? |
| 22:17.06 | WIMPy | No, But they will start production, as soon, as you place your order. |
| 22:17.09 | WIMPy | ~ask |
| 22:17.09 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:19.02 | JoseBravo | In other words, is possible to change the codec to established SIP channel? |
| 22:19.34 | WIMPy | AFAIK both in theory and practice. |
| 22:19.42 | *** join/#asterisk ShaunR (~shaun@freenode/sponsor/NDChost.com) |
| 22:21.13 | ShaunR | i have a polycom 550 phone behind a linksys router that had 2 phone lines out with sip options inside (disabled). I ditched that router and got a new one and now my polycom 550 phone is not working. I had this problem before when trying to swap this router out a while back with another linksys unit without the voip stuff on it. The new router has no firewall enabled... anybody know what |
| 22:21.13 | ShaunR | might be going on. |
| 22:22.51 | *** join/#asterisk QbY (~kelvin@96.176.19.11) |
| 22:23.26 | diemos | ShaunR: where is the router in relation to your PBX or SIP trunk? |
| 22:23.41 | ShaunR | sorry, inbetween (internet) |
| 22:24.36 | diemos | so the phones operate off of two POTS lines and a SIP trunk as well? |
| 22:24.46 | QbY | When using RFC2833, is it possible to change the "Volume" level? |
| 22:25.03 | ShaunR | no, the phones make a sip connection over the internet to the asterisk server.. |
| 22:25.15 | ShaunR | phone -> nat -> router -> internet -> asterisk |
| 22:26.02 | diemos | ShaunR: Do you have them provisioned or just adding the extensions manually via the web interface on the phone? |
| 22:26.10 | WIMPy | QbY: There is no volume. The Information os not sent as tones. |
| 22:26.39 | QbY | WIMPy: There is a field in the packet that says volume.. |
| 22:26.47 | ShaunR | diemos: they where working fine until i swapped out the router, nothing other than the router changed. |
| 22:27.09 | QbY | WIMPy: And one of our termination providers is stating the volume is probably the reason we have so many DTMF recognition problems |
| 22:27.16 | WIMPy | QbY: Interesting. Well, maybe some gateways will make use of it. |
| 22:27.17 | diemos | ShaunR: Try putting one of the phones in a DMZ and see if that works |
| 22:28.20 | WIMPy | QbY: So you're not using RFC2833? |
| 22:28.42 | QbY | WIMPy: Yes, I am using RFC2833. |
| 22:29.00 | WIMPy | QbY: So where does the volume come in? |
| 22:29.20 | WIMPy | Describe the whole setup. |
| 22:29.45 | ShaunR | diemos: nope, not working |
| 22:29.45 | QbY | let me dump one of these packets for you |
| 22:30.12 | WIMPy | That won't help. |
| 22:30.41 | WIMPy | Describe the whole path on which you're having issues. |
| 22:31.55 | QbY | Handset (both Polycom, and Cisco) -> Asterisk -> Carrier |
| 22:33.02 | WIMPy | And how are you connected to that carrier? |
| 22:33.45 | delroy | Anyone using analog TDM cards in their setups? |
| 22:34.27 | WIMPy | delroy: Unless you ask a real question, you won't have any luck in getting an answer. |
| 22:36.08 | delroy | Looking for opinions on quality. What are recommendations from the following: Sangnoma, Digium, Rhino?, and HW echo cancellation or no hw echo cancellation? Will software echo cancellation cut it? |
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| 22:37.36 | QbY | WIMPy: We are connected via SIP/Internet |
| 22:38.17 | WIMPy | QbY: Ok, and how are you transmitting dtmp to (or from?) your provider? |
| 22:38.34 | WIMPy | ftmf |
| 22:38.41 | WIMPy | argl |
| 22:38.43 | WIMPy | dtmf |
| 22:39.49 | diemos | ShaunR: I would go and manually reconfigure each of the phones. That's about all I can say without being on your network >< |
| 22:40.39 | diemos | Otherwise, I'd say MITM between a phone and the gateway to see what kind of traffic is coming out. You can usually pick up registration tries and get a little bit more detail from sniffing packets. |
| 22:42.38 | QbY | WIMPy: RFC2833 |
| 22:42.57 | citywok | hahaha, http://i.imgur.com/A7mbk.jpg |
| 22:44.05 | WIMPy | QbY: If you use RFC2833 all the way there's nothing that can go wrong. Are you sure they support RFC2833? |
| 22:44.50 | QbY | they say they do.. he just came back at me and told me that his volume is set at 13, where ours is saying 10 |
| 22:46.49 | WIMPy | Thet volume can only be of any interest if the infomation is converted to tones, i.e. on a media gateway. |
| 22:47.15 | WIMPy | As long as the call stays sip there is not much room for interpretation. |
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| 22:48.58 | QbY | I understand that, however when we call certain companies they aren't recognize DTMF when terminating with this carrier |
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| 22:51.38 | QbY | WIMPy: Take a look at http://pastebin.com/G6yWLjS8 (1.1.1.1 = Phone, 2.2.2.2 = Asterisk, 3.3.3.3 = Carrier) |
| 22:52.05 | QbY | WIMPy: Take a look at this specifically: "..00 1010 = Volume: 10" -- That's what they want increased. |
| 22:52.15 | WIMPy | Ah, so it's not what you described but something beyond? |
| 22:52.27 | WIMPy | What your provider then converts to the PSTN? |
| 22:53.02 | QbY | Yes |
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