| 00:04.44 | TJNII | reads the question, reads the username, reads the question again... |
| 00:04.55 | TJNII | I'm guessing this has to be fairly robust and reliable? |
| 00:05.17 | Yudaisrael1984 | has to be somewhat reliable yes |
| 00:05.17 | Kyosh | heheh |
| 00:06.35 | TJNII | I don't know of any packages that will do that, but you should be able to make one with SQL and call files pretty quick. |
| 00:06.46 | TJNII | I'd say one day of coding if you know SQL. |
| 00:07.11 | *** join/#asterisk The-Bat (~The-Bat@122.182.0.38) |
| 00:07.15 | TJNII | This is also assuming that you have enough numbers to make SQL advantageous. |
| 00:08.29 | Kyosh | yuda, i know of services that do this, but not software |
| 00:09.20 | Yudaisrael1984 | im trying not to reinvent the wheel so i thought maybe theres some open source progect out there but i dont even know how to call it in order to search for it |
| 00:10.52 | TJNII | Heck, you could probably do it in a bash "one liner." By "one liner" I mean one of those "on liners" that is technically one line, but wraps to 8 or 10 lines on the terminal. |
| 00:12.38 | Yudaisrael1984 | thanks but what would this be called? |
| 00:15.50 | manxpower | "phone spam" |
| 00:22.52 | Naikrovek | manxpower: heh |
| 00:23.06 | Naikrovek | record the audio you want to go out to each callee |
| 00:23.21 | Naikrovek | write a script to generate a bunch of asterisk call files, which play that audio file |
| 00:23.27 | Naikrovek | move the call files to the outgoing directory |
| 00:23.36 | Naikrovek | asterisk will make the calls and play the file, then hang up |
| 00:24.10 | Naikrovek | if you use this to tell everyone about your newest product offering, you will likely get visited by an assassin in the middle of the night |
| 00:25.21 | TJNII | Make sure the move to the outgoing directory is atomic. |
| 00:25.33 | Naikrovek | yes |
| 00:25.35 | Naikrovek | all or nothing |
| 00:25.42 | TJNII | If memory serves Asterisk can be ... aggressive ... when it comes to call files. |
| 00:25.56 | Naikrovek | move the files from a directory on the same partition or you'll have trouble |
| 00:26.36 | Naikrovek | the VERY CLOCK CYCLE that the call file appears, asterisk will read the call file, whether you've finished writing the call file or not |
| 00:26.42 | Naikrovek | so don't just write them to the outgoing directory |
| 00:26.58 | Naikrovek | write them to another directory on the same filesystem then MOVE them to the outgoing directory |
| 00:27.05 | TJNII | How does Asterisk do that, anyways? |
| 00:27.18 | Naikrovek | system events i'm assuming |
| 00:27.35 | Naikrovek | you can monitor a directory then the OS will tell you when the file appears |
| 00:27.37 | TJNII | Huh. I didn't know there were hooks for that. |
| 00:27.43 | TJNII | Learn something new everyday. |
| 00:28.03 | Naikrovek | wel |
| 00:28.04 | Naikrovek | well |
| 00:28.05 | Corydon76-dig | Naikrovek: Not until trunk it won't |
| 00:28.07 | Kyosh | im thinking even using perl scripts to drive AGI's to launch the calls maybe even from a manager interfasce |
| 00:28.29 | Naikrovek | cannot parse "not until trunk" |
| 00:28.29 | TJNII | I really think call files would be easier. |
| 00:28.37 | Corydon76-dig | Naikrovek: 1.8 |
| 00:28.44 | Naikrovek | Corydon76-dig: ah |
| 00:29.06 | Corydon76-dig | Presently, the method is a busy loop |
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| 00:29.13 | Naikrovek | Corydon76-dig: oh you're kidding |
| 00:29.15 | Naikrovek | wow |
| 00:29.47 | Corydon76-dig | Starting in 1.8, pbx_spool will use inotify on Linux, kqueue on BSD, and the old busy loop for everything else |
| 00:30.00 | Corydon76-dig | Thank you, POSIX, for no standard method. |
| 00:30.15 | Naikrovek | in C# on mono you can set up an event hook that monitors a directory for changes. the INSTANT something in that directory changes, you are notified and can take action in the event handler |
| 00:30.34 | Naikrovek | (That runs on linux, so i'm assuming it's a kernel hook you can grab on to) |
| 00:31.16 | Corydon76-dig | On Linux, the kernel method is called inotify |
| 00:31.18 | Naikrovek | and the only reason i know this is because i ran into the "i'll notify you before you're done writing the file" |
| 00:31.25 | Naikrovek | in my little app |
| 00:32.17 | Naikrovek | thank you, Corydon76-dig |
| 00:32.22 | Corydon76-dig | Once POSIX creates a common abstraction layer for it, we'll use that, but..... |
| 00:32.31 | Naikrovek | learning is funs |
| 00:33.02 | Corydon76-dig | In some ways, inotify is a better interface than kqueue, though kqueue does more |
| 00:34.37 | Corydon76-dig | and kqueue has some really fucked concepts, which led Apple to create some weird flags to get around them |
| 00:35.12 | Corydon76-dig | but even Apple's implementation is neither complete nor fully standard |
| 00:36.07 | florz | Corydon76-dig: you mean, like, 1.8 will also fix the race condition in the call file stuff? |
| 00:36.42 | Corydon76-dig | florz: do you have a reproducable case for the race condition? |
| 00:37.23 | Corydon76-dig | If you can reliably reproduce the race, that is the key element that prevented me from putting it in 1.4 and other releases. |
| 00:37.58 | Corydon76-dig | Because if I cannot show that it makes a difference, then it's just a feature, not a bugfix |
| 00:39.33 | florz | Corydon76-dig: no, never tried to, just read the source |
| 00:40.28 | Corydon76-dig | I agree that there's theoretically a race condition, but I need to be able to reproduce it in a test |
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| 00:42.15 | florz | that's a strange definition of a bug, if you ask mee :-) |
| 00:42.27 | florz | -e |
| 00:43.08 | Corydon76-dig | florz: I need to be able to reproduce it, so I can show that using inotify makes a difference |
| 00:43.30 | Corydon76-dig | otherwise, it's not a fix |
| 00:44.04 | florz | Corydon76-dig: yeah I understood what you said - it's still a strange policy, if you ask me |
| 00:44.28 | Corydon76-dig | Why? |
| 00:44.57 | florz | if it's provably incorrect code (as in: is provably not guaranteed to do what it's supposed to do), it's a bug? |
| 00:45.30 | Corydon76-dig | florz: If it's provable incorrect code, then you should have no problem reproducing the bug |
| 00:45.41 | Corydon76-dig | right? |
| 00:45.44 | florz | erm ... no? |
| 00:45.57 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:46.27 | florz | that's essentially the nature of concurrency bugs - incorrect code does not necessarily lead to easily reproducable incorrect behaviour |
| 00:46.47 | WIMPy | loved the 80s when there was something called the robustness principle. - Those where the days... |
| 00:47.49 | Corydon76-dig | WIMPy: Right, which means that we don't rewrite code that works, unless you can demonstrate that the rewrite is for an actual bug |
| 00:48.43 | florz | or more practically: concurrency bugs usually are the source for "general instability and unreliability you don't really know of where it comes from" |
| 00:48.44 | WIMPy | Corydon76-dig: You just wrote that there is a race condition. That fits my definition of a bug. |
| 00:48.58 | Corydon76-dig | WIMPy: there is theoretically a race condition |
| 00:49.17 | Corydon76-dig | WIMPy: but if it never actually manifests, then it's not actually a bug |
| 00:49.37 | WIMPy | Corydon76-dig: Our definitions differ. |
| 00:50.14 | florz | the point being that you simply don't know whether it actually manifests - all you know is that it does not manifest often enough for people to figure out what the source of the problem is |
| 00:51.06 | Corydon76-dig | florz: find me someone who has the problem on a regular basis, even if they cannot reproduce it at will |
| 00:51.32 | Corydon76-dig | Then give them the patch. If the problem disappears, that works for me |
| 00:52.10 | florz | now, you are changing your own definition - a few moments ago, it was "actually", now it is "regularly" |
| 00:52.16 | Corydon76-dig | but I gotta have a "this patch fixed the problem for me" |
| 00:52.38 | Corydon76-dig | florz: this isn't a definition game |
| 00:53.12 | Corydon76-dig | If you want to play word games, there's plenty of other places on the Internet for you to be |
| 00:56.19 | florz | well, you started playing it, didn't you? |
| 00:56.57 | florz | you started changing the requirements for considering something a bug, I pointed out you did so |
| 00:57.27 | florz | and I don't agree with either |
| 00:58.16 | Corydon76-dig | You're more than welcome to take it up on the -dev list and ask that the inotify support on pbx_spool be backported to 1.4 |
| 00:59.12 | Corydon76-dig | If there's no objections (like the kind that we get from people who are understandably upset when we change implementations unnecessarily), I'll backport it myself |
| 01:00.07 | Corydon76-dig | but the whole idea behind policy is to avoid breaking things unnecessarily (even if I believe in my heart of hearts that no such breakage should occur) |
| 01:00.49 | Corydon76-dig | I have been shown to be wrong in the past about possible breakage, and I'm sure it'll happen again in the future, though not for lack of trying otherwise |
| 01:02.52 | florz | well, yeah, that part I totally do understand - but for that it's really more a matter of finding the balance between getting rid of bugs and avoiding unintended side effects of bug fixes, which nonetheless would be bug fixes for real bugs |
| 01:03.16 | Corydon76-dig | That is precisely my point |
| 01:03.54 | Corydon76-dig | If it ain't a bug that's affecting anybody, then it ain't worth the danger of possibly breaking the branch |
| 01:05.42 | florz | and part of that IMO is not just a matter of how bad the bug seems to be in practical terms (which is a bit difficult to judge, after all), but also how likely the fix is to break stuff - after all, there are bug fixes where you can be pretty sure you won't break anything, and fixes where it's a bit difficult to know for sure what the side effect are |
| 01:06.35 | Corydon76-dig | The problem is that fixes that I thought were in the former case can and do turn out to be the latter case |
| 01:07.00 | florz | Well, yeah, I'd think that that's bad luck if it doesn't happen too often ;-) |
| 01:07.18 | Corydon76-dig | This is again why we have policy |
| 01:08.20 | Corydon76-dig | Find me someone where the patch improves their production systems, and I have something I can run with. |
| 01:09.02 | Corydon76-dig | Otherwise, it's my ass on the line for no actual benefit |
| 01:10.46 | Corydon76-dig | I hate to be that crass, but trying to be polite wasn't working. |
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| 02:58.30 | p3nguin | Is there a setting to change the voicemail access key from * to something else? I'm talking about the key to press to get into VoiceMailMain when already in a user mailbox. |
| 03:00.56 | ChannelZ | haxx0r the source |
| 03:02.33 | p3nguin | I'll take that as "it's not user-configurable." |
| 03:04.05 | ChannelZ | not that I know of. pretty much all of the keys are hard-coded in |
| 03:05.18 | p3nguin | Some voicemail systems use the # key, some require pressing 9... now people are going to have to remember that Comedian Mail uses the star key. Way to standardize. |
| 03:22.48 | Kyosh | comedian |
| 03:22.50 | Kyosh | love that name |
| 03:32.38 | coppice | I think it was named after the developer of the original meridian mail |
| 03:32.56 | p3nguin | Because he was a comedian? |
| 03:33.25 | coppice | how else would he have come up with the design of meridian mail? |
| 03:33.35 | p3nguin | :) |
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| 04:46.20 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 05:04.22 | manxpower | p3nguin: 1.6.mumble has the MiniVm system |
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| 05:07.00 | TJNII | Haha, I read that as "1.6.mumble has the MiniVan system" |
| 05:07.09 | TJNII | Asterisk 1.6: Now for Soccer Moms! |
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| 05:52.24 | ChannelZ | heh |
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| 07:06.18 | *** join/#asterisk ZenBSDi (~danny@unaffiliated/ZenBSDi) |
| 07:06.32 | ZenBSDi | Hello room. |
| 07:06.44 | ZenBSDi | Anyone know whats the call scaling performance of 1.6? |
| 07:06.49 | ZenBSDi | I can't find any numbers |
| 07:07.10 | tzafrir_laptop | For starters, "1.6" is not a version of asterisk |
| 07:07.22 | tzafrir_laptop | 1.6.0, 1.6.1 and 1.6.2 are separate branches |
| 07:07.27 | ZenBSDi | Asterisk 1.6.2 |
| 07:07.44 | tzafrir_laptop | And then again, could you please define "call scaling"? |
| 07:08.06 | ZenBSDi | I thought all 1.6 uses the new hash memory system per call to increase call concurrency |
| 07:08.10 | tzafrir_laptop | Different workloads behave differently |
| 07:12.56 | ZenBSDi | tzafrir_laptop: I'm talking about this. |
| 07:12.58 | ZenBSDi | The only rule of thumb we appear to be able to provide is this: Asterisk 1.2 start to run into problems around 220 concurrent SIP calls. Asterisk 1.4 scales much better and can handle nearly double the call setups/second as well as total concurrent traffic. Moreover, Early testing of Asterisk 1.6 using hash tables shows a SIP performance increase compared to 1.4 of factor 3 to 4. |
| 07:13.08 | ZenBSDi | http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 07:13.47 | tzafrir_laptop | ZenBSDi, it depends very much on the hardware, the type of transocding, and other things |
| 07:13.56 | tzafrir_laptop | Test yourself |
| 07:14.20 | tzafrir_laptop | Either using sipp, or using a number of other Asterisk servers |
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| 08:13.24 | bennyt | need advice on recording issues, anyone able to help? |
| 08:17.26 | bennyt | I'm having issues with only one side of calls being recorded when using monitor |
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| 08:24.48 | ChannelZ | I'm tired but don't want to go to bed. |
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| 09:13.23 | ido- | Hi, can anyone please look at this error: http://pastebin.org/175928 |
| 09:16.50 | c0rnoTa | you're receive "Forbidden" in replay for your INVITE packet from telco as I see. So, try to request explanation from your telco why they reject your request. |
| 09:17.38 | c0rnoTa | it could be wired with re-registrations from different ports if you behind the nat |
| 09:17.54 | ido- | I'm not behind nat |
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| 09:18.33 | ido- | and they don't have official sip support (not using their softphone or hardware), so I won't get an answer for such a question. |
| 09:18.39 | lmns972 | hello everybody |
| 09:18.45 | lmns972 | I have a problem to configure asterisk with my provider (free) |
| 09:18.54 | ido- | the thing is, that it works fine once i reset the device (until some time passes) |
| 09:18.59 | lmns972 | Everything seems okay but the incoming and outgoing calls do not work. |
| 09:19.06 | lmns972 | if you have a idea |
| 09:19.16 | lmns972 | http://pastebin.org/175941 |
| 09:21.30 | c0rnoTa | ido-: look in your sip packet transfer schema. You send them INVITE, and receive Forbidden. They didn't sent you "Proxy Auth required". |
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| 09:22.37 | ChannelZ | lmns972: you show no evidence that the calls don't work |
| 09:23.41 | ido- | c0rnoTa: what does that mean ? |
| 09:23.53 | lmns972 | ChannelZ, I'm new to asterisk, how active your logs? |
| 09:24.52 | ChannelZ | What do you see in the console when you make a call, or try to get one? |
| 09:24.57 | ChannelZ | core set verbose 4 |
| 09:25.11 | c0rnoTa | ido-: That mean that now they rejects your requests for a one of reasons. |
| 09:25.40 | lmns972 | ChannelZ, ok i try for the moment there are nothing |
| 09:25.50 | ido- | for what reasons ? |
| 09:25.51 | c0rnoTa | ido-: it could be registration timeout or different outgoing ports |
| 09:25.58 | ido- | registraion timeout.. |
| 09:26.07 | ido- | haven't thought of that ! |
| 09:26.32 | c0rnoTa | ido-: and much more reasons what they could imagine |
| 09:28.02 | lmns972 | ChannelZ, When I make a outgoing call I get this mesage "== Using SIP RTP CoS mark 5" |
| 09:28.20 | ido- | Ok, i'll try lowering register expires timeout |
| 09:28.29 | ido- | i think i saw 1800 on their original hardware |
| 09:28.46 | c0rnoTa | ido-: wish you luck ã |
| 09:28.51 | ChannelZ | lmns972: you should get more than that - in the console type 'core set verbose 4' |
| 09:30.47 | lmns972 | ChannelZ, I typed core set verbose 4, I make a call and I get the message I gave you |
| 09:31.59 | ChannelZ | then it's going through |
| 09:32.03 | ChannelZ | somewhere anyway |
| 09:33.15 | ChannelZ | although generally it should be showing some call progress (or a rejection) after that |
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| 09:33.55 | *** join/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 09:34.00 | DelphiWorld | hi all |
| 09:34.03 | DelphiWorld | anyone from the uk? |
| 09:34.09 | DelphiWorld | i need someone that is bt client |
| 09:34.13 | DelphiWorld | bt==britich telecom |
| 09:34.50 | lmns972 | ChannelZ, I almost was not the log |
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| 09:35.59 | coppice | BT = Bad Telecoms |
| 09:36.18 | DelphiWorld | coppice: ;) |
| 09:36.33 | DelphiWorld | coppice: fs==free services |
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| 09:36.48 | lmns972 | ChannelZ, I think I'm having a problem with modules |
| 09:37.00 | ChannelZ | lmns972: I'm going to guess your dialplan is not setup or something, as you should be seeing other things after '== Using SIP RTP CoS mark 5'.. like a Dial command executing namely |
| 09:39.52 | lmns972 | ChannelZ, how to activate the dialplan? there is a module? |
| 09:41.40 | ChannelZ | Didn't you setup anything in extensions.conf ? |
| 09:42.22 | lmns972 | yes |
| 09:42.30 | lmns972 | it's all |
| 09:43.44 | lmns972 | ChannelZ, what is sip logs can be useful? |
| 09:46.39 | ChannelZ | sorry what you're saying doesn't make any sense nor does it add up. The console should be revealing an error or something past what you have said, and without knowing any of that, I can't help you |
| 09:46.41 | ChannelZ | I'm going to bed |
| 09:47.03 | DelphiWorld | ChannelZ: good night;) |
| 09:47.43 | lmns972 | ChannelZ, ok |
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| 11:00.37 | maja100 | Hi guys I wonder if you can help me, I've follwed the steps on the yum installation and I get the following back from my terminal |
| 11:00.47 | maja100 | login as: root |
| 11:00.47 | maja100 | root@94.229.66.169's password: |
| 11:00.47 | maja100 | root@x [~]# yum install asterisk16 asterisk16-configs asterisk16-voicemail dahdi-linux dahdi- |
| 11:00.47 | maja100 | Loaded plugins: fastestmirror |
| 11:00.47 | maja100 | Loading mirror speeds from cached hostfile |
| 11:00.48 | maja100 | <PROTECTED> |
| 11:00.48 | maja100 | <PROTECTED> |
| 11:00.49 | maja100 | <PROTECTED> |
| 11:00.49 | maja100 | <PROTECTED> |
| 11:00.50 | maja100 | Excluding Packages in global exclude list |
| 11:00.50 | maja100 | Finished |
| 11:00.51 | maja100 | Setting up Install Process |
| 11:00.51 | maja100 | Parsing package install arguments |
| 11:00.52 | maja100 | No package asterisk16 available. |
| 11:03.27 | maja100 | hello.... |
| 11:04.24 | maja100 | can anyone help with my query? |
| 11:09.50 | maja100 | how do I gey any help on here? |
| 11:12.16 | *** part/#asterisk maja100 (~mark@host81-158-103-230.range81-158.btcentralplus.com) |
| 11:12.28 | *** join/#asterisk maja100 (~mark@host81-158-103-230.range81-158.btcentralplus.com) |
| 11:13.30 | maja100 | Is anyone on here available to help me with my Asterisk yum installation? |
| 11:15.35 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-155-51.cinci.res.rr.com) |
| 11:15.59 | *** join/#asterisk imcdona (~t@valium.voicebyip.net) |
| 11:17.39 | maja100 | hello.... |
| 11:20.42 | tzafrir_laptop | maja100, hi |
| 11:21.25 | tzafrir_laptop | infobot, tell maja100 about pb |
| 11:22.06 | tzafrir_laptop | maja100, what linux distribution do you use? |
| 11:22.19 | maja100 | CentOS |
| 11:22.21 | maja100 | 5 |
| 11:22.35 | tzafrir_laptop | Those packages are not in standard repositories |
| 11:22.44 | tzafrir_laptop | have you added any extra repositories? |
| 11:23.06 | maja100 | Basicaly I'm a newbie, I've follwed everything on the Asterisk website |
| 11:24.29 | maja100 | I'm using WinSCP to add repos files |
| 11:24.40 | maja100 | sorry .repo |
| 11:27.40 | *** join/#asterisk Z_God (~julius@wlan239224.mobiel.utwente.nl) |
| 11:32.54 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 11:33.43 | maja100 | how do i ask for help on here? |
| 11:34.26 | jaytee | ~ask |
| 11:34.27 | 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. |
| 11:36.47 | tzafrir_laptop | maja100, just pastebin that file |
| 11:37.21 | maja100 | http://pastebin.com/LqHwT025 |
| 11:46.28 | tzafrir_laptop | maja100, I meant: the .repo file |
| 11:47.49 | maja100 | tzafrir_laptop, which one would you like me to pastebin? |
| 11:48.05 | tzafrir_laptop | <maja100> I'm using WinSCP to add repos files |
| 11:48.28 | tzafrir_laptop | What repositories have you added (or tried to add) ? |
| 11:49.33 | maja100 | centos-asterisk.repo |
| 11:49.46 | maja100 | centos-digium.repo |
| 11:50.01 | maja100 | I've added both as per the Asterisk guide |
| 12:09.09 | maja100 | tzafrir_laptop, any ideas? |
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| 12:10.02 | tzafrir_laptop | maja100, where have you added them to? |
| 12:10.19 | tzafrir_laptop | wonders if yum has issues with CR-LF text files |
| 12:12.15 | maja100 | tzafrir_laptop, I added them to: /etc/yum.repos.d |
| 12:28.01 | maja100 | tzafrir_laptop, just a update the base url in the .repo files was slight wrong so I've updated so tried installing again and thi is what i get |
| 12:28.02 | maja100 | http://pastebin.com/0GMsLcmF |
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| 12:28.39 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:33.36 | tzafrir_laptop | maja100, looks like a minor packaging bug |
| 12:33.52 | tzafrir_laptop | Not sure which files are supposed to be as the user asterisk |
| 12:35.15 | maja100 | tzafrir_laptop, any ideas where I can support to correct it? |
| 12:36.27 | tzafrir_laptop | maja100, is asterisk running? try: asterisk -r |
| 12:39.14 | maja100 | tzafrit_laptop, done that and here is what it came back with: http://pastebin.com/pn332Rhd |
| 12:48.38 | jaytee | ~sipnat |
| 12:48.38 | infobot | [~sipnat] Quick guide on configuring * + 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 |
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| 13:12.12 | tzafrir_laptop | maja100, looks like it is not running |
| 13:15.10 | maja100 | tzafrir_laptop, it looks like it's running now, what should I do next? http://pastebin.com/8f3u8UWu |
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| 13:40.55 | af_ | could I compile asterisk 1.4.30 against h323plus? |
| 13:47.23 | tzafrir_laptop | af_, have you tried? |
| 13:47.39 | af_ | I am trying now |
| 13:48.15 | tzafrir_laptop | af_, does h323plus use pkg-config or does it include a openh323-config script? |
| 13:49.06 | af_ | it looks not |
| 13:50.27 | tzafrir_laptop | which of the two? a .pc file or a -config script? |
| 13:52.03 | maja100 | tzafrir_laptop, sorry to bother you again how do I log in to the web GUI I have cpanel and WHM also installed on the server so would it be via a port? |
| 13:52.49 | tzafrir_laptop | maja100, that's rather off-topic . it's normally port 2083 or something similar |
| 13:53.01 | af_ | tzafrir_laptop, none of them |
| 13:55.34 | tzafrir_laptop | af_, hmm... that's odd |
| 13:55.53 | tzafrir_laptop | does the configure script detect it? |
| 13:56.06 | af_ | \uhm, no |
| 13:56.38 | af_ | I have set the env vars, like pwlib and openh323 |
| 13:58.35 | af_ | now with -- args on configure |
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| 14:08.51 | maja100 | anyone know which port number I use to log in to the web gui? |
| 14:11.29 | fenrus | what web gui ? |
| 14:11.46 | fenrus | that probably depends on your configuration anyways |
| 14:12.39 | Pimmetje | try nmap your system to see what ports are open :) |
| 14:13.40 | maja100 | pimmetje, thanks |
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| 14:23.40 | af_ | uhm found some documentation |
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| 14:46.59 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 14:47.04 | eject_ck | Hi |
| 14:48.16 | eject_ck | I'm trying to use * to send fax using callfile and sip chanel where provider supports t.38 |
| 14:48.16 | eject_ck | I'm getting outgoing call to fax on other side, but then asterisk crashes |
| 14:48.33 | eject_ck | what I see in logs related to this these 3 lines http://pastebin.com/JXFTmPq1 |
| 14:49.05 | eject_ck | Found unknown media description format X-NSE (what does it mean?) |
| 14:50.48 | coppice | NSE = network signaling event - see RFC4733 and RFC4734 I don't know what the X- part would imply |
| 14:52.06 | eject_ck | coppice: does it matter in this case ? |
| 14:52.26 | coppice | its very unlikely |
| 14:53.23 | eject_ck | hm |
| 14:53.26 | eject_ck | thanks |
| 14:53.38 | eject_ck | how to catch why asterisk crashes ? |
| 14:53.46 | eject_ck | nothing usefull in console |
| 14:55.06 | eject_ck | <PROTECTED> |
| 14:55.07 | eject_ck | <PROTECTED> |
| 14:55.07 | eject_ck | Disconnected from Asterisk server |
| 14:55.07 | eject_ck | Executing last minute cleanups |
| 14:55.13 | eject_ck | it's what I see |
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| 14:59.54 | LemensTS | can i force a user to rergister |
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| 15:18.35 | Bartockbatz | hello folks - anyone have time to answer a question? |
| 15:20.26 | Bartockbatz | almost afraid to ask but I was wondering if someone knew a free/opensource/less expensive alternative to Visual Dialplan Pro - I am on a tight budget and cannot really drop the coin on it |
| 15:20.44 | eject_ck | how figure what minimal list of modules I need to send faxes via sip |
| 15:23.57 | Bartockbatz | almost afraid to ask but I was wondering if someone knew a free/opensource/less expensive alternative to Visual Dialplan Pro - I am on a tight budget and cannot really drop the coin on it |
| 15:24.01 | Bartockbatz | anyone? |
| 15:24.07 | Bartockbatz | Beuller? Beuller? |
| 15:29.37 | Gugge | Bartockbatz: i know a completely free alternative .... dont use it, and write the dialplay yourself :) |
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| 15:30.50 | Gugge | is there something like a str_replace function, to replace a string in a variable? |
| 15:31.25 | Bartockbatz | good plan - on that note, being a novice, any good resources for writing dialplans? |
| 15:32.45 | bennyt | I'm having issues with only one side of calls being recorded when using monitor, anyone able to help? |
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| 15:40.56 | af_ | tzafrir_laptop, it is compiling chanH323 now |
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| 16:00.21 | PwrSurge | hi |
| 16:00.31 | PwrSurge | what's the best linux distro to use for Asterisk? |
| 16:00.50 | bennyt | centos is pretty normal i can't say the best |
| 16:02.03 | [TK]D-Fender | PwrSurge: Whichever you are capable of managing |
| 16:02.21 | [TK]D-Fender | PwrSurge: * doesn't care so long as the pre-reqs are installe |
| 16:02.23 | [TK]D-Fender | d |
| 16:02.26 | PwrSurge | ok |
| 16:02.34 | PwrSurge | so even Ubuntu would be fine? |
| 16:02.41 | [TK]D-Fender | PwrSurge: As I said... |
| 16:02.52 | PwrSurge | cool |
| 16:03.53 | [TK]D-Fender | PwrSurge: what other roles will that machine be serving? |
| 16:05.36 | PwrSurge | light web surfing and email for a web based POS terminal |
| 16:06.52 | lost_soul | asterisk now uses centos if memory serves |
| 16:06.54 | [TK]D-Fender | PwrSurge: and you're going to run your server on it? |
| 16:07.23 | lost_soul | I use asterisk on openbsd personally, have also ran it on freebsd.. works nice |
| 16:07.33 | PwrSurge | yes |
| 16:07.47 | [TK]D-Fender | PwrSurge: as long as you keep your loads low you should be fine |
| 16:08.23 | PwrSurge | the POS terminal server is on another box though |
| 16:08.30 | p3nguin | <lost_soul> asterisk now uses centos if memory serves <-- No, Asterisk uses whatever you deploy it on. |
| 16:08.31 | manxpower | ~answers |
| 16:08.32 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 16:08.33 | manxpower | ~mailinglist |
| 16:08.34 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 16:09.20 | bennyt | trixbox uses it |
| 16:09.33 | PwrSurge | so basically just gnome and install firefox and evolution mail client |
| 16:09.44 | lost_soul | p3nguin: I meant the live CD called "AsteriskNow" |
| 16:10.15 | lost_soul | sorry for the confusion, I should have typed it correctly |
| 16:10.31 | [TK]D-Fender | lost_soul: s'ok, he's now our resident grammar AND spacing nazi. |
| 16:10.48 | lost_soul | lol |
| 16:11.44 | [TK]D-Fender | Your abs are lute leer write! |
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| 16:12.34 | bennyt | anyone aware of why I can only hear one side of the convo when recording on TDM? |
| 16:18.39 | manxpower | bennyt: what are you using to record? |
| 16:21.00 | bennyt | well i'm using asterisk-java to use the manager API |
| 16:21.11 | bennyt | calling the monitor action |
| 16:25.20 | PwrSurge | if I install the AsteriskNOW ISO, does it also install the Gnome desktop as well? |
| 16:25.22 | [TK]D-Fender | bennyt: Whats the name of the file you are playing back? |
| 16:25.54 | [TK]D-Fender | PwrSurge: No, you would have to install that yourself after |
| 16:27.11 | PwrSurge | ok |
| 16:27.22 | PwrSurge | so it's based on centos right? |
| 16:32.49 | [TK]D-Fender | PwrSurge: Yes |
| 16:34.51 | *** part/#asterisk maja100 (~mark@host81-158-103-230.range81-158.btcentralplus.com) |
| 16:37.54 | bennyt | [TK]D-Fender: the one that works is filename-in.wav and the one that is always at 44 bytes is filename-out.wav |
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| 16:53.45 | bennyt | any ideas [TK]D-Fender? |
| 16:54.33 | [TK]D-Fender | bennyt: What is on the other side of the call? |
| 16:54.47 | bennyt | the agent |
| 16:55.00 | bennyt | so the agent is on the recording but the customer isn |
| 16:55.01 | bennyt | 't |
| 16:55.53 | bennyt | but like i said, when the agent uses SIP or IAX it works just fine |
| 16:56.15 | bennyt | I'm hoping its just some sort of configuration maybe? |
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| 17:13.38 | bennyt | suggestions [TK]D-Fender? |
| 17:15.43 | [TK]D-Fender | no idea |
| 17:15.54 | [TK]D-Fender | bennyt: what ver of *? |
| 17:16.08 | bennyt | v1.6 |
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| 17:18.29 | bennyt | nothin eh? :( |
| 17:18.41 | [TK]D-Fender | bennyt: that isn't a version |
| 17:21.05 | bennyt | Asterisk 1.6.0.10-FONCORE-r40 |
| 17:21.37 | [TK]D-Fender | bennyt: Considerably out of date. Update time. |
| 17:22.33 | bennyt | intersting, ok. just a "yum update" ? |
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| 17:33.47 | [TK]D-Fender | bennyt: Was * installed by yum? |
| 17:39.10 | manxpower | bennyt: packages are not supported here. The packagers usually screw something up. |
| 17:40.12 | bennyt | yeah thats what I've been reading |
| 17:40.21 | bennyt | running yum update blows things ups |
| 17:40.23 | bennyt | up* |
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| 17:40.39 | bennyt | i'm using trixbox too so it's probably built with dependencies on its current snapshot |
| 17:41.02 | bennyt | manxpower: do you know why i might be having this problem? |
| 17:42.32 | bennyt | the frustrating part is that it works when using IAX and SIP but we just can't use em |
| 17:49.00 | TJNII | ~trixbox |
| 17:49.01 | infobot | rumour has it, trixbox is SH1TB0X. Basically a CRAPPY, closed source distro. STAY AWAY! |
| 17:49.47 | bennyt | agreed. we tried getting asterisknow up but with the whole dahdi switch, the cards were causing major problems |
| 17:50.22 | manxpower | ~freepbx |
| 17:50.23 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 17:59.48 | manxpower | Trixbox is not closed source and it's not a distro |
| 17:59.59 | manxpower | 'well, at least not closed source, I'm not sure about the distro part |
| 18:11.13 | moy | Qwell: there? |
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| 19:16.45 | Kobaz | do de do |
| 19:22.01 | ChannelZ | doodie |
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| 19:49.16 | bennyt | btw, I figured out the issue with my channels (well kind of hacked) |
| 19:49.59 | bennyt | calling mixmonitor vs monitor does something that works for TDM |
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| 20:53.05 | sun28 | moin |
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| 20:58.04 | ChannelZ | bleh |
| 21:01.55 | sun28 | blah |
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| 21:13.19 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:13.23 | *** join/#asterisk Geminizer (~gatekeepe@cpe-76-180-18-183.buffalo.res.rr.com) |
| 21:14.01 | *** part/#asterisk Geminizer (~gatekeepe@cpe-76-180-18-183.buffalo.res.rr.com) |
| 21:17.24 | *** join/#asterisk boodu (~boodu@175.158.129.128) |
| 21:19.45 | boodu | hello |
| 21:35.34 | ChannelZ | hi |
| 21:58.51 | *** join/#asterisk Peste_Bubonica (~eduardo@189.44.254.3) |
| 21:58.51 | Peste_Bubonica | Hi all |
| 22:01.56 | manxpower | ~mailinglist |
| 22:01.58 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 22:03.09 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:05.21 | Peste_Bubonica | is ok to have this output from dahdi /proc? Span 1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: Linux26) 1" (MASTER) |
| 22:05.28 | *** join/#asterisk QaDeS (~mklaus@p4FC73723.dip0.t-ipconnect.de) |
| 22:05.33 | Peste_Bubonica | the source is not to be some timer? |
| 22:05.35 | Peste_Bubonica | like rtc? |
| 22:15.30 | *** join/#asterisk blaines (~blaines@ip70-190-67-139.ph.ph.cox.net) |
| 22:16.54 | *** join/#asterisk blaines (~blaines@ip70-190-67-139.ph.ph.cox.net) |
| 22:18.35 | *** join/#asterisk blaines (~blaines@ip70-190-67-139.ph.ph.cox.net) |
| 22:27.44 | *** join/#asterisk blaines (~blaines@ip70-190-67-139.ph.ph.cox.net) |
| 22:31.13 | *** join/#asterisk blaines (~blaines@ip70-190-67-139.ph.ph.cox.net) |
| 22:44.24 | *** join/#asterisk xpot-mobile (~james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 22:44.25 | *** join/#asterisk riddlebox (~riddlebox@75-132-219-116.dhcp.stls.mo.charter.com) |
| 23:03.03 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:18.02 | *** join/#asterisk root52 (~root52@wsip-70-183-187-149.cl.ri.cox.net) |
| 23:18.50 | root52 | Good day all. Question. Is there a way I can read the SIP responce into the dialplan? |
| 23:29.05 | *** join/#asterisk QbY (~kelvin@c-24-126-145-123.hsd1.ga.comcast.net) |
| 23:29.40 | QbY | http://pastebin.com/ARFWTg8z -- ConfBridge is crashing my entire box when people hangup. Doesn't happen everytime, but enough. |
| 23:29.44 | manxpower | root52: You cannot do that using only Asterisk. You CAN get access to the SIP headers. |
| 23:30.09 | root52 | "The internet" has left me doubtful. but i bassicaly want to do diffrent thing based on the sip responce. ie tell the diffrence between a 404 and another code. I looked at the whole HANGUPCAUSE but it seem to always return "0" regardless of the SIP codes. |
| 23:31.29 | root52 | manxpower: Thanks. So what would I need to "get" access to the SIP header. I can see the SIP responce using the CLI or tcpdump but I need to figure out how to manulipate the dialplan based on that. Perhaps an AGI script or something? |
| 23:33.28 | manxpower | root52: read channelvariables.tex or channelvariables.txt or the asterisk.pdf (all should be in the asterisk doc/ directory) or check the voip-info.org wiki for the channel variable info, just remember the wiki is frequently out of date and sometimes just plain wrong |
| 23:34.04 | root52 | manxpower: Thanks a bunch. |
| 23:34.06 | manxpower | HANGUPCAUSE should be set to whatever Asterisk maps the SIP response to. |
| 23:34.33 | manxpower | There is an RFC for SIP/Q.931 cause code mapping |
| 23:35.03 | *** join/#asterisk hipitihop (~denis@203.132.229.236) |
| 23:35.42 | root52 | I saw that. And am good with the translitions. I "caused" a 404 to be returned and then a call that went thru. I can see diffrent responces but HANGUPCAUSE alwasy seems to return 0 |
| 23:37.04 | root52 | manxpower I will read the txt file and hack around with HANGUPCAUSE for a bit and see where that gets me. Thanks. |
| 23:41.13 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |