| 00:00.07 | rob0 | I'm going to cross compile (in a way) a Linux x86 binary on a Linux x86_64 machine. Not true cross compiling because I' |
| 00:00.32 | rob0 | ll be in a chroot with an x86 OS. Any known gotchas for * ? |
| 00:01.40 | rob0 | Should something like "export ARCH=i386" (or whatever) do? |
| 00:03.26 | docelmo | Strom_C yes that should work |
| 00:03.42 | Strom_C | docelmo: excellent |
| 00:03.42 | docelmo | Cause your giving a reason to the network as to why you couldnt terminate the call |
| 00:03.59 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 00:04.02 | Strom_C | right, my only question was whether PRI_CAUSE gets sent backwards along the IAX trunk |
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| 00:04.15 | Strom_C | obviously it will work on the box that has the T1 card |
| 00:04.21 | docelmo | sorry Im working on a HUGE technical document for a Class4/Class5 switch development out of Asterisk and FreeSwitch |
| 00:04.43 | Strom_C | no problem |
| 00:05.00 | docelmo | Yes I believe that bit is set. I havent looked at the core of the IAX stack but I believe it gets set in there somewhere |
| 00:05.50 | docelmo | My company was gonna dump low-mid 6 figures into software that would have NEVER done the job they wanted it to do.. So now I get to hire a development team and make it happen |
| 00:06.02 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 00:06.19 | Strom_C | sweet |
| 00:06.40 | rob0 | $OSCPU ... but hmmm ... not covered in the ifeq |
| 00:06.48 | docelmo | so I am building a Class 4 Tandem (wholesale switch) and a Class 5 (end user switch) out of it. |
| 00:07.15 | Strom_C | gasp, I cant believe you think I don't know the difference between a class 4 and a class 5 ;) |
| 00:07.33 | rob0 | Class 1! |
| 00:07.45 | Strom_C | 404-1 |
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| 00:09.11 | docelmo | Well you are asking basic PRi questions abotu asterisk |
| 00:09.18 | docelmo | its not you I do it to everyone |
| 00:09.40 | Strom_C | heh, ok...it was more a "finer points of IAX" question than a PRI question |
| 00:11.48 | docelmo | iax isnt bad.. it has its hiccups |
| 00:12.14 | rob0 | oh well ... going for a walk before it gets too dark. I'll try to compile this later. |
| 00:14.02 | docelmo | have fun with that |
| 00:14.22 | docelmo | I have to come up with a scope of work.. :) |
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| 00:27.57 | i-ball | okay, so icecast is installed and so is ices2 |
| 00:28.00 | i-ball | question |
| 00:28.04 | i-ball | This tutorial: |
| 00:28.15 | i-ball | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ICES |
| 00:28.32 | i-ball | references the file: "Modify asterisk/contrib/asterisk-ices.xml as needed. " |
| 00:28.44 | i-ball | Where is asterisk-ices.xml? I can't find it anywhere |
| 00:29.01 | Strom_C | /usr/src/asterisk/contrib etc |
| 00:29.41 | Strom_C | and also, it's generally not necessary to preface a question with the word "question" when suffixing it with a question mark will work just fine :) |
| 00:30.19 | i-ball | heh, good point. |
| 00:33.00 | i-ball | I'm not seeing it in /usr/src/asterisk/contrib |
| 00:33.19 | Strom_C | which version of asterisk are you running? |
| 00:33.55 | i-ball | 1.2.7.1 |
| 00:34.34 | Strom_C | first off, upgrade to 1.2.10. second, the file is in the contrib subdirectory of the asterisk source directory |
| 00:34.49 | i-ball | and 1.2.9 patch 1 on a different machine |
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| 00:41.43 | mitcheloc | hey russellb were you familliar with the res_jabber/chan_jingle stuff in asterisk? |
| 00:41.43 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
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| 00:42.30 | russellb | mitcheloc: sort of .... |
| 00:42.51 | russellb | it has been a couple months since i have looked at it |
| 00:43.14 | file | hi |
| 00:43.16 | mitcheloc | i'm getting a really strange message -> http://pastebin.ca/89806 |
| 00:44.40 | russellb | looks like it can't find libiksemel on your system :) |
| 00:44.45 | russellb | ls /usr/lib/libiksemel* |
| 00:45.23 | russellb | however, that shouldn't happen ... because it won't be allowed to build unless it could be found ... |
| 00:46.09 | mitcheloc | russellb, right, it built it, i downloaded "iksemel-1.2.tar.gz" and did a ./configure - make - make install, so i'm a bit confused |
| 00:46.33 | russellb | i ... don't know |
| 00:46.44 | mitcheloc | i don't see it in /usr/lib |
| 00:46.52 | russellb | so where did it get installed? |
| 00:48.44 | mitcheloc | russellb, http://pastebin.ca/89810 -- looks like in /usr/local/lib maybe... |
| 00:49.01 | mitcheloc | yep it's in there |
| 00:49.19 | russellb | ugh, so if it went there, how did the configure script find it? |
| 00:49.22 | russellb | the world may never know |
| 00:49.45 | russellb | if you're lazy ... mv /usr/local/lib/libiksemel* /usr/lib |
| 00:49.58 | russellb | or, actually. |
| 00:50.01 | russellb | even better .... |
| 00:50.13 | mitcheloc | i'm running a clean minimal install of centos, so nothing on my system should have been out of the ordinary |
| 00:50.15 | russellb | go back to libiksemel, and do ./configure --prefix=/usr && make && make install |
| 00:50.42 | russellb | that should put it where it needs to be |
| 00:52.16 | mitcheloc | russellb, okay it looks like it loaded it now, cool, should this issue i had be documented/fixed anywhere? |
| 00:52.38 | russellb | i'm not sure how to classify it |
| 00:52.55 | russellb | because the way autoconf works, it literally builds a program against the library and makes sure it runs successfully |
| 00:53.04 | russellb | to make sure it's actually usable |
| 00:53.09 | russellb | so that situation shouldn't ever happen |
| 00:53.13 | russellb | ... in theory |
| 00:53.28 | russellb | so i really don't know what the issue is ... and i'm probably too tired to think about it :) |
| 00:53.35 | *** join/#asterisk setti (n=setti@32-76-112.adsl.terra.cl) |
| 00:53.51 | mitcheloc | heh, well it figures that thats just my luck that i get impossible problems =/ |
| 00:57.48 | Dovid | sdf |
| 00:58.26 | docelmo | Say whats the FCC thing for monitoring calls in the US Callea? How's it spelled? |
| 00:58.40 | mitcheloc | it sounds like a virus |
| 00:59.53 | docelmo | nevermind.. got it |
| 01:00.01 | Strom_C | CALEA |
| 01:00.03 | *** join/#asterisk SarahEmm (n=sarahemm@MTL-HSE-ppp159791.qc.sympatico.ca) |
| 01:00.14 | russellb | mitcheloc: well, we got it working ... so whatever :) |
| 01:00.15 | Strom_C | the Communications Assistance for Law Enforcement Act or somesuch |
| 01:00.19 | SarahEmm | hihi |
| 01:03.02 | mitcheloc | russellb, true :) now to configure it! |
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| 01:18.40 | Idle | the volume on my voicemails that are attached to emails is quite low... like, excessively low... is there a way to equalize that to be a LOT louder? |
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| 01:29.55 | MRH2 | anyone using a dell 2650? |
| 01:31.23 | Skarmeth | OT: Anyone know how to make destar work? Where I can set it's Manager username/password? |
| 01:33.45 | *** join/#asterisk SarahEmm (n=sarahemm@MTL-HSE-ppp159791.qc.sympatico.ca) |
| 01:35.12 | mitcheloc | now the jabber bits are complaining that gnuTLS isn't installed, "yum install gnutls" seemed to do it, but it is not being dected |
| 01:35.16 | mitcheloc | *detected |
| 01:40.40 | file | probably need the development package |
| 01:43.50 | Strom_C | Idle: what kind of connection are your calls coming in on? |
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| 01:50.30 | _Vile | using the manager api with action: originate and using "varible: name=value" -- ${name} is not available in that context after the call is originated, am i doing something wrong? |
| 01:50.40 | _Vile | err variable: |
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| 01:52.43 | asteriskAnew | how do the ports get selected with a tdm2400 to a 66 block punch down |
| 01:52.44 | asteriskAnew | ?? |
| 01:52.47 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 01:52.59 | Strom_C | asteriskAnew: standard 25-pair color code mappings |
| 01:56.27 | Strom_C | asteriskAnew: or is that not what you're asking |
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| 02:04.47 | Strom_C | asteriskAnew: or did you just come here to ask a question and then ignore the response? |
| 02:05.25 | benjk | maybe he had to run to the bathroom or something like that |
| 02:05.40 | Strom_C | for twelve minutes? |
| 02:05.43 | benjk | oh well |
| 02:05.51 | Strom_C | :) |
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| 02:16.25 | *** join/#asterisk NoRemorse (n=bah@210-84-10-183.dyn.iinet.net.au) |
| 02:16.27 | NoRemorse | hi all |
| 02:16.47 | Setti | hi |
| 02:16.56 | NoRemorse | how do I get * to dial 1 extension after the other rather than all at same time please? ie Dial(SIP/401&SIP/402) |
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| 02:17.30 | SarahEmm | NoRemorse: just use seperate Dial() statements i think |
| 02:17.50 | rob0 | sure, different priorities |
| 02:18.14 | NoRemorse | cant do it in 1 command? |
| 02:18.44 | SarahEmm | don't think so |
| 02:19.02 | NoRemorse | ok |
| 02:19.49 | dlynes_laptop | SarahEmm, how's T.O.? |
| 02:20.38 | Strom_C | NoRemorse: yeah, just have one dial statement time out and then continue to the next one |
| 02:21.00 | SarahEmm | dlynes_laptop: too hot, too humid. do i know you? |
| 02:21.13 | dlynes_laptop | well, you've seen me on here |
| 02:21.17 | dlynes_laptop | but that's it |
| 02:21.37 | dlynes_laptop | I'm in Vancouver |
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| 02:21.48 | SarahEmm | okay :) |
| 02:21.54 | dlynes_laptop | so, no |
| 02:21.56 | SarahEmm | it's okay. overwarm tho |
| 02:21.57 | dlynes_laptop | I'm not stalking you :) |
| 02:22.01 | SarahEmm | okay ;) |
| 02:22.38 | dlynes_laptop | yeah...been cooking here lately |
| 02:22.48 | dlynes_laptop | I don't miss southern ontario summers, whatsoever |
| 02:23.49 | SarahEmm | oh cool, someone finally came out with a SIP overhead paging speaker :) |
| 02:24.09 | dlynes_laptop | where? |
| 02:24.28 | SarahEmm | http://www.cyberdata.net/voip/voip-speaker.html |
| 02:24.43 | SarahEmm | still no standalone ringers out that i can see. i need to get going on that.. |
| 02:24.50 | NoRemorse | thanks |
| 02:25.31 | dlynes_laptop | nice |
| 02:25.41 | dlynes_laptop | paging speaker is still useful, though |
| 02:28.25 | feld_ | sweet paging speaker. |
| 02:28.32 | feld_ | we were looking for some. i'm gonna be ordering.... |
| 02:28.33 | feld_ | :) |
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| 03:13.41 | TomRob | question about the API, anyone feel like giving an asterisk developer working late on a sunday night a bit of help? |
| 03:14.07 | TomRob | anyone alive? |
| 03:15.21 | Strom_C | alive possibly |
| 03:15.22 | _Vile | what about the api? |
| 03:15.27 | Strom_C | helpful maybe |
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| 03:18.15 | TomRob | heh, ok, here goes, trying to do a click to call using the API from a webpage, the issue is when I do a click to call to SIP phone to PSTN, works easy, It dials the SIP phone, when it picks up goes for the outbound call |
| 03:18.53 | TomRob | well, I need to do PSTN to PSTN, the only way I found to do this so far is using Local/PhoneNumber insted of the SIP phone and it dials both at the same time |
| 03:19.25 | TomRob | I would like to get back to the first, where it calls the first PSTN line, if it picks up, makes the call to the other PSTN |
| 03:19.32 | TomRob | what im doing just seems like a hack |
| 03:19.33 | h3x0r | Local/ channel is broken |
| 03:20.15 | h3x0r | use manager |
| 03:20.34 | SplasPood | h3x0r: huh? |
| 03:20.50 | TomRob | I was about to say them same. huh? |
| 03:21.17 | h3x0r | Local is broken |
| 03:21.19 | TomRob | Local/PhoneNumber works,, exp. Local/18005551212 |
| 03:21.26 | h3x0r | it has a lot of bugs in it |
| 03:21.30 | h3x0r | it makes asterisk crash |
| 03:21.32 | SplasPood | works fine for me... |
| 03:21.34 | SplasPood | never did that... |
| 03:21.37 | SplasPood | and either way |
| 03:21.41 | SplasPood | what does 'manager' solve? |
| 03:21.52 | SplasPood | Presumably he's using either the manager API or call files |
| 03:22.02 | TomRob | Im using the API |
| 03:22.04 | SplasPood | the two are mutually exclusive.. (Manager and Local/) |
| 03:22.09 | Strom_C | TomRob: are you using POTS lines or PRI? |
| 03:23.13 | TomRob | Im using PRI, Im connecting to another provider to make the calls, (voipjet) |
| 03:23.32 | Strom_C | so in reality you're using IAX |
| 03:23.44 | Strom_C | PRI-by-proxy :) |
| 03:23.47 | TomRob | yep |
| 03:23.53 | TomRob | ;) |
| 03:24.27 | Strom_C | what happens when you just use Dial(IAX2/voipjet/number) instead of using chan_local for the calls? |
| 03:26.21 | TomRob | Well, cant use that.. reason is there is a small AGI script that needs to make the call to track them.. |
| 03:26.27 | TomRob | using that bypasses it |
| 03:27.44 | Strom_C | and the AGI can't call Dial? |
| 03:28.22 | TomRob | yes, it can, I just need to pass it the number to dial, that is how im doing it now using the Local/phonenumber |
| 03:29.11 | Strom_C | so instead of doing local/phonenumber, why not either use iax2/voipjet/phonenumber or fix your damned agi script? |
| 03:30.45 | Strom_C | and what kind of fancy tracking does your agi do that can't be handled by call detail record parsing? |
| 03:31.35 | TomRob | heh, its doing call rating |
| 03:32.09 | TomRob | I would just like to get the same effect as the SIP to PSTN were it calls the SIP phone first, If picks up, calls the PSTN line |
| 03:32.50 | TomRob | I just cant seem to get that effect when using the PSTN to PSTN other then doing the direct dial from the API as Strom_C said |
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| 03:33.10 | Strom_C | so why not rate the call separately? |
| 03:33.24 | Strom_C | have an AGI that rates the call and returns a value |
| 03:34.47 | TomRob | not sure I understand, how can I call the AGI if im doing the direct dial from the API? |
| 03:35.08 | Strom_C | I don't think I understand your program's structure |
| 03:35.14 | Strom_C | walk me through the whole thing from start to finish |
| 03:36.43 | TomRob | ok, the goal of the project is to create bridge 2 PSTN calls, At this time, using the API, if you do SIP to PSTN it will first call the SIP ext, wait for an answer, then dial the PSTN |
| 03:36.55 | Strom_C | no, thats not what im asking for |
| 03:37.11 | Strom_C | walk me through the steps the ACTUAL CODE takes to set the whole thing up |
| 03:37.26 | TomRob | ok, 1min, let me get it... |
| 03:37.43 | Strom_C | what? |
| 03:37.54 | Strom_C | you dont conceptually know what your program is supposed to do? |
| 03:38.04 | Strom_C | how are you supposed to debug it then? :) |
| 03:38.17 | TomRob | I was getting you the code im using |
| 03:38.30 | Strom_C | i just want you to explain it to me step by step |
| 03:38.38 | Strom_C | i dont want to actually debug your code |
| 03:38.59 | Strom_C | I want to understand what your code does so that I can help you figure out where the problem is |
| 03:39.11 | TomRob | ok, well.. not much code to debug anyway, just an api call, and it works ( to a point) I will take it step by step.. |
| 03:40.10 | TomRob | ok, someone goes to a website at school, enters then phone number on a webpage, in the backgound it makes an API call to dial the teacher, if he picks up, it then makes another dial to the person on the website that entered the phone number |
| 03:41.04 | TomRob | this all works now.. then only issue I have is that its making both calls at the same time. calls the teacher and the person on the website at the same time |
| 03:41.31 | Strom_C | sigh |
| 03:41.39 | Strom_C | detail, man |
| 03:41.41 | Strom_C | what calls what |
| 03:41.52 | Strom_C | what are the specific steps you take |
| 03:42.00 | Strom_C | how does call flow progress |
| 03:42.05 | Strom_C | etc etc etc etc |
| 03:42.11 | TomRob | I first telnet into the asterisk box |
| 03:42.11 | Nugget | telnet is eeeeeeevil! |
| 03:42.38 | TomRob | then use the org command to call both numbers |
| 03:43.56 | Strom_C | what is the exact dial string you're using? |
| 03:43.59 | TomRob | on the Channel: i use Local/PSTN Number exp.. Local/7774546545 |
| 03:44.36 | TomRob | on the Exten: I use what the person entered into the webpage 7445545541 |
| 03:44.49 | Strom_C | why not do it this way |
| 03:45.13 | Strom_C | have the program first call the originating party |
| 03:45.27 | Strom_C | then, once the party answers, perform a second call setup to the destination party |
| 03:47.29 | TomRob | when it first calls the Originating party, what would you enter for a Channel? |
| 03:48.56 | Strom_C | use Local/whatever or IAX/whatever |
| 03:50.08 | Strom_C | then have the extension on the box be an AGI which handles authenticating the user and placing the second leg of the call |
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| 03:51.40 | TomRob | In the AGI, I would need somthing that would first check to see if the Orginating party picked up befor it makes the call?? that the idea? |
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| 03:52.45 | Strom_C | you say something like "please press two to be connected" |
| 03:52.51 | Strom_C | wait for a two |
| 03:52.53 | Strom_C | and then place the call |
| 03:53.04 | Strom_C | hell, you dont even need an AGI for that |
| 03:53.12 | Strom_C | just do it in dialplan logic :) |
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| 03:54.32 | TomRob | ok, im with you,, but I would still need to pass the phone number to be called |
| 03:54.46 | Strom_C | sure; you have it go into its own context |
| 03:55.39 | TomRob | yep, that would work, im kind of doing that now, only thing is from that context, im just calling the rating AGI and making the call,, mabye just put a small dialplan like you said.. |
| 03:55.49 | Strom_C | so have it walk through storing that number into a channel variable, perform an answer and confirmation and whatnot, then "2" dials the channel variable |
| 03:55.58 | Strom_C | why must you use your rating AGI? |
| 03:56.17 | TomRob | well, I need to keep track of the "costs" of calls |
| 03:56.38 | SplasPood | why not go through your CDRs and calc cost later? |
| 03:56.58 | wunderkin | that would be too easy |
| 03:57.03 | SplasPood | wunderkin: :) |
| 03:57.12 | Strom_C | exactly - that's not the kind of thing you should worry about at runtime |
| 03:57.18 | Strom_C | billing is a batch process :) |
| 03:57.21 | TomRob | becouse the work has already been done on the AGI |
| 03:57.26 | SplasPood | umm |
| 03:57.34 | SplasPood | thats not the "right" way to do it.. |
| 03:57.36 | wunderkin | agi es teh suc |
| 03:57.40 | SplasPood | I think thats what people are trying to tell you |
| 03:57.43 | SplasPood | AGI is not teh suck |
| 03:57.48 | SplasPood | its just not the right tool for this job |
| 03:57.59 | SplasPood | And anyway |
| 03:58.04 | SplasPood | how do you calculate the cost of a call |
| 03:58.08 | Strom_C | TomRob: and your laziness has gotten you a broken piece of code :) |
| 03:58.09 | SplasPood | before you know how long the call is? |
| 03:58.16 | i-ball | wow |
| 03:58.41 | i-ball | icecast documentation leaves a bit to be desired for.. or however the saying goes |
| 03:59.14 | TomRob | well, if you dont have credit or allowed to place the call, it will not let you.. if you do have credit and place the call, if you go over on that call, you will not be able to make any other calls |
| 03:59.48 | Strom_C | TomRob: that's balance checking, not call rating |
| 04:00.12 | TomRob | well, its all in one, it gets the price of the call, check the balance and places the call |
| 04:00.22 | Strom_C | and apparently it isn't working for you |
| 04:00.22 | SplasPood | whats the price of the call |
| 04:00.22 | TomRob | records the costs in the database, etc.. etc.. |
| 04:00.28 | SplasPood | wouldn't that vary with the length? |
| 04:00.36 | Strom_C | SplasPood: yeah, thats what i'm wondering |
| 04:01.41 | TomRob | yes, for example, you have 5 credits, a call to guam costs 1 credit a min.. it will allow you to make this call, but will warn you when you are running down to 1 credit |
| 04:01.45 | TomRob | then kill the call |
| 04:01.53 | TomRob | and not allow you to place and other calls |
| 04:02.15 | Strom_C | glad you told this to us at the beginning of the conversation |
| 04:02.16 | TomRob | its a whole system, I did not write it, just the luck one that has to "add" on to it.. :) |
| 04:02.43 | SplasPood | Hrm, how do you inject audio into a call? |
| 04:02.58 | TomRob | its to stop abuse. not sure, I did not write it as I said |
| 04:03.10 | CunningPike | i-ball: It leaves a lot to be desired :) |
| 04:03.13 | SplasPood | yea, that was a general question for all :) |
| 04:04.14 | TomRob | I did notice that when the msg comes on, you have 1min left or credit that it does seem to stop the call for a min, mabye its putting both people on hold, playing that msg to one of the partys, then resuming the call |
| 04:04.15 | TomRob | just a guess |
| 04:04.55 | Strom_C | TomRob: does it work if, just for shits and giggles, you bypass the AGI and dial the destination party directly? |
| 04:05.15 | TomRob | no, the AGI is not called |
| 04:05.22 | TomRob | so you can bypass it |
| 04:05.38 | Strom_C | um |
| 04:05.43 | TomRob | if I used a Dial(Blahh,Blahh) from the API |
| 04:05.44 | Strom_C | that answer makes no sense |
| 04:05.56 | TomRob | this would bypass it |
| 04:07.26 | Strom_C | right |
| 04:07.29 | Strom_C | what i'm asking is: |
| 04:07.57 | h3x0r | i am sick and tired of clueless people doing calling card and toll bypass systems that arent licensed |
| 04:08.22 | Strom_C | if you bypass the AGI and just do two dial statements from your AMI program, does the supervision protocol work as expected? |
| 04:08.40 | SplasPood | this is an AGI? |
| 04:08.44 | SplasPood | you sure this isn't |
| 04:08.49 | SplasPood | app_somecustomthing.so |
| 04:09.11 | Strom_C | he claims it's an AGI anyway |
| 04:09.33 | SplasPood | Strom_C: yea, that was directed towards TomRob |
| 04:09.48 | Strom_C | i know |
| 04:09.57 | *** join/#asterisk AJay-mn (i=TJBoi@70.59.126.196) |
| 04:10.36 | adfsadfae | anyone know how I can get freepbx to work with vicidialer |
| 04:10.41 | TomRob | Its making a call to DeadAGI and running an AGI script |
| 04:10.51 | Strom_C | why DeadAGI? |
| 04:10.56 | Strom_C | adfsadfae: read the topic |
| 04:12.31 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 04:12.33 | Corydon76-home | h3x0r: who would they license it from? |
| 04:12.46 | adfsadfae | i meant how would I configure an installation of multiple servers of asterisk consisting of separate mysql, php, asterisk? |
| 04:12.56 | TomRob | hmm, that I dont know Strom_C, I did not make it, it does somthing like DeadAGI,somescript.php |
| 04:13.04 | Corydon76-home | adfsadfae: very carefully |
| 04:13.18 | SplasPood | bahaha |
| 04:13.24 | Strom_C | TomRob: just for shits and giggles, try AGI() instead of DeadAGI() |
| 04:13.25 | SplasPood | that was a great about face |
| 04:13.35 | adfsadfae | .... |
| 04:13.39 | Strom_C | haha yes, that should win a prize |
| 04:13.55 | TomRob | before I crash and burn the system, what should that do for me Strom_C? |
| 04:13.59 | TomRob | :) |
| 04:14.07 | Strom_C | I need some sort of automatic web-based certificate generator :) |
| 04:14.14 | Strom_C | TomRob: please tell me this isn't a production system |
| 04:14.52 | TomRob | just used for myself and a project im working on |
| 04:15.01 | Strom_C | TomRob: give it a shot and tell me if it works |
| 04:18.42 | *** join/#asterisk asterisk-dud (n=dwwollma@64-42-247-120.mb.skyweb.ca) |
| 04:19.04 | asterisk-dud | anyone know y my tdm400p fxs card doesnt hang up? |
| 04:19.12 | TomRob | Strom_C, well, I just did a simple call using it with AGI, worked |
| 04:19.17 | Strom_C | asterisk-dud: didnt we go over that? |
| 04:19.34 | asterisk-dud | no |
| 04:19.40 | Strom_C | asterisk-dud: what do you mean "doesnt hang up" |
| 04:19.59 | Strom_C | TomRob: good |
| 04:20.02 | Strom_C | TomRob: problem solved |
| 04:20.15 | Strom_C | TomRob: go read up on the differences between deadagi and agi :) |
| 04:20.25 | asterisk-dud | lets say i dial in on it, then once both ends hang up, asterisk keeps the line active until my timeout hang it up |
| 04:20.33 | TomRob | ahh, well I know the DeadAGI is after a hang up |
| 04:20.37 | TomRob | correcT? |
| 04:20.43 | Strom_C | asterisk-dud: FXS port? |
| 04:20.46 | Strom_C | asterisk-dud: or FXO port? |
| 04:20.46 | asterisk-dud | yes |
| 04:20.50 | asterisk-dud | FXS |
| 04:20.51 | TomRob | or mabye I need to get out my book again |
| 04:21.05 | Strom_C | TomRob: also go read up on answer supervision |
| 04:21.20 | TomRob | k |
| 04:21.26 | Strom_C | asterisk-dud: how is it configured |
| 04:21.57 | asterisk-dud | fxoks |
| 04:22.15 | Strom_C | ok, and what do you mean by "once both ends hang up"? |
| 04:22.28 | asterisk-dud | lets say u call me and hang up |
| 04:22.35 | Strom_C | also, what kind of station equipment are you using on that line? |
| 04:22.53 | asterisk-dud | before i answer, the phone cont ringing until i pick up, then i get a busy signal |
| 04:23.01 | hads|home | So you are actually talking about the FXO port. |
| 04:23.06 | h3x0r | an IXC license is required for calling card applications |
| 04:23.17 | Strom_C | asterisk-dud: is the call coming in over the fxo port? |
| 04:23.19 | asterisk-dud | the fxo side hagns up |
| 04:23.43 | asterisk-dud | it comes is on fxs and asterisk forwards the line into an fxo channel |
| 04:24.03 | hads|home | h3x0r: You're talking about the US. |
| 04:24.04 | Strom_C | asterisk-dud: since the fxo port never supervises, asterisk has no way of knowing that the line has finished ringing until it realizes that the ringing signal isn't happening anymore |
| 04:24.23 | Strom_C | h3x0r: really? calling cards are patented now? :) |
| 04:24.45 | asterisk-dud | is there any way to solve this? |
| 04:24.55 | Strom_C | asterisk-dud: don't use an FXO line? :) |
| 04:25.06 | TomRob | h3x0r I will be sure to do down the 7-11 and pick one up :) |
| 04:25.18 | shmaltz | h3x0r, I believe only for calling card on mobile phones applications (like prepaid) |
| 04:25.20 | asterisk-dud | well all my internal extentions are fxo |
| 04:25.24 | Strom_C | asterisk-dud: or answer the call immediately as it comes in |
| 04:25.30 | hads|home | Or get disconnect supervision from the telco |
| 04:25.46 | Strom_C | hads|home: disconnect supervision is useless if the call is never connected in the first place |
| 04:26.07 | shmaltz | Strom_C, not true |
| 04:26.26 | hads|home | Here you still get an open loop on disconnect even if the call isn't answered. |
| 04:26.27 | *** part/#asterisk asterisk-dud (n=dwwollma@64-42-247-120.mb.skyweb.ca) |
| 04:26.31 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 04:26.36 | Strom_C | hads|home: I'm talking about on inbound calls |
| 04:26.38 | shmaltz | Strom_C, you still want to be able to know if the phone stopped ringing or not |
| 04:26.43 | Corydon76-home | Yay, analog! |
| 04:26.46 | hads|home | So am I |
| 04:26.57 | docelmo | oi! |
| 04:27.01 | Strom_C | hads|home: I have never seen the telco stop ringing accompanied by a battery drop |
| 04:27.37 | hads|home | I am in .nz so things could be wildly different :) |
| 04:27.45 | Strom_C | yes, things probably are |
| 04:27.49 | Strom_C | I'm in .us |
| 04:28.32 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:29.48 | Strom_C | you can't dodge...THE POWER OF BAKELITE!! |
| 04:31.38 | Qwell | clearly.. |
| 04:31.50 | Strom_C | hey Qwell :) |
| 04:31.56 | Qwell | y0 |
| 04:32.03 | Strom_C | what cooks? |
| 04:32.20 | Qwell | apparently, the high desert, heh |
| 04:32.26 | Strom_C | apart from me in my apartment without air conditioning |
| 04:32.35 | Strom_C | my thermostat was maxed out earlier |
| 04:32.43 | Strom_C | needle was all the way to the right |
| 04:33.21 | Qwell | nice |
| 04:33.34 | shmaltz | Strom_C, there is a little thing that can help you about that needle problem |
| 04:33.43 | Qwell | How's the weather been this past week? |
| 04:34.00 | Strom_C | Qwell: hot and a bit muggy during the day |
| 04:34.02 | shmaltz | Strom_C, just take the cover off, and twist it from the inside clockwise, about halfway, that should fix it |
| 04:34.05 | cHr1Zt1An | hello, where I can find information about asterisk for beginners ?? |
| 04:34.09 | Strom_C | but fantastic at night |
| 04:34.11 | Qwell | Strom_C: excellent |
| 04:34.12 | Strom_C | cHr1Zt1An: |
| 04:34.14 | Strom_C | ~bok |
| 04:34.17 | Strom_C | er |
| 04:34.18 | Strom_C | ~book |
| 04:34.19 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:34.19 | shmaltz | ~docs |
| 04:34.20 | jbot | well, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 04:34.31 | cHr1Zt1An | thanks |
| 04:35.29 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 04:35.37 | Strom_C | Qwell: hows the weather in west covina? |
| 04:35.50 | Qwell | Strom_C: not sure yet :p |
| 04:35.57 | Strom_C | oh? |
| 04:36.00 | Strom_C | where are you? |
| 04:36.03 | Qwell | home now |
| 04:36.06 | Strom_C | alabama? |
| 04:36.10 | Strom_C | where were you? |
| 04:36.10 | Qwell | was |
| 04:36.14 | Strom_C | cool |
| 04:37.30 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 04:38.11 | hads|home | Speaking of bakelite, anyone see that old rotary with a GSM module installed? I found it amusing. |
| 04:38.29 | Strom_C | hads|home: yes. They used an Automatic Electric set. Cretins!!! |
| 04:52.25 | *** join/#asterisk }btorch{ (n=kvirc@c-66-176-87-59.hsd1.fl.comcast.net) |
| 04:52.41 | }btorch{ | hey anyone here usiing wildfire with asterisk-IM |
| 04:56.00 | docelmo | not yet.. but SOON! |
| 04:56.24 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001217014b92.vc.shawcable.net) |
| 04:56.29 | Strom_C | I want Rick Ocasek Echo Cancellers! |
| 04:56.37 | Strom_C | er |
| 04:56.38 | Strom_C | Ric |
| 04:57.08 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-159.range86-139.btcentralplus.com) |
| 04:57.21 | Strom_C | obviously my right hand middle finger doesnt read liner notes from my record albums :) |
| 04:57.26 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 04:58.14 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001217014b92.vc.shawcable.net) |
| 05:01.46 | Strom_C | *crickets( |
| 05:01.54 | Strom_C | * not ( |
| 05:01.58 | docelmo | anyone interested in a IM client that does text, iax, video and TV? |
| 05:02.09 | docelmo | err IPTV? |
| 05:02.24 | docelmo | and is fully jabber compatible :) |
| 05:02.30 | Strom_C | docelmo: I would be if I didn't have disdain for television and softphones |
| 05:02.48 | docelmo | disdain? |
| 05:02.59 | sugardave | I need my tv for console games |
| 05:03.21 | Strom_C | docelmo: I haven't owned a TV set since 2002 |
| 05:03.36 | docelmo | ohh no you can keep your TV we will have set top boxes too :) |
| 05:04.42 | docelmo | Good lord.. I would die without my 40" TV and my 32" LCD monitor |
| 05:05.13 | Strom_C | television bores the living dogballs out of me |
| 05:05.26 | sugardave | oh, the shows are crap, for sure |
| 05:05.34 | docelmo | I dont know.. I love house |
| 05:05.46 | docelmo | I cant get enough.. He reminds me alot of well me.. |
| 05:05.57 | Strom_C | I was once operatrd on by a Dr. House |
| 05:06.06 | Strom_C | so of course i expect the show to be about ear surgery |
| 05:06.21 | docelmo | Nope.. diagnostic medacine |
| 05:06.34 | Strom_C | six points lost for misspelling medicine |
| 05:06.49 | sugardave | cool, there's a scorecard? |
| 05:06.49 | Strom_C | also, six points lost to me for not correcting operatrd |
| 05:06.52 | Strom_C | yes |
| 05:07.05 | docelmo | Im a engineer I dont have to spell.. Just make this shit work |
| 05:07.14 | sugardave | where can I download mine so I can play at home? |
| 05:07.25 | docelmo | huh? |
| 05:07.29 | hads|home | It's 26-4 at the moment. |
| 05:07.29 | Strom_C | sugardave: its not downloadable; its on a whiteboard on my wall :) |
| 05:07.35 | sugardave | nice |
| 05:07.41 | Strom_C | next to the payphones |
| 05:08.53 | docelmo | payphones? |
| 05:09.01 | Strom_C | I have payphones in my apartment |
| 05:09.07 | docelmo | why? |
| 05:09.13 | docelmo | you dont have a landline? |
| 05:09.16 | Strom_C | yes I do |
| 05:09.27 | Strom_C | they're connected to my asterisk box |
| 05:09.43 | i-ball | does it use ACTS though? |
| 05:09.48 | Strom_C | i-ball: not yet |
| 05:09.53 | docelmo | ahh well.. Im out.. lata.. |
| 05:09.58 | _Vile | <PROTECTED> |
| 05:10.08 | docelmo | yes |
| 05:10.11 | _Vile | CDR(duration) and CDR(billsecs) don't get set until hangup |
| 05:10.17 | Strom_C | i-ball: the GTE AE120 acts like a regular phone, and i have yet to program rate tables into the protel |
| 05:10.18 | docelmo | ohh live? |
| 05:10.22 | _Vile | :) |
| 05:10.23 | docelmo | not that I know of |
| 05:10.31 | Strom_C | _Vile: easy |
| 05:10.42 | Strom_C | _Vile: set a variable with epoch time at the start of the call |
| 05:10.52 | Strom_C | _Vile: then do comparisons as necessary |
| 05:11.13 | *** join/#asterisk asterisk-dud (n=dwwollma@64-42-247-120.mb.skyweb.ca) |
| 05:11.35 | _Vile | ahh ok |
| 05:11.36 | _Vile | thx |
| 05:11.57 | Strom_C | ${EPOCHTIME} or ${EPOCH} will contain the data you need |
| 05:12.03 | Strom_C | i forget which it is |
| 05:12.05 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-159.range86-139.btcentralplus.com) |
| 05:12.44 | _Vile | rockin' |
| 05:16.39 | Corydon76-home | ${EPOCH} |
| 05:17.12 | Corydon76-home | and ${TIMESTAMP} and ${DATETIME} are both deprecated |
| 05:20.13 | *** join/#asterisk jetaway2009 (n=asd@60.48.167.61) |
| 05:20.20 | jetaway2009 | hi |
| 05:27.19 | *** join/#asterisk linlin (n=linlin@c-67-184-159-30.hsd1.il.comcast.net) |
| 05:37.25 | *** join/#asterisk dlynes_ (n=dlynes@S0106001217014b92.vc.shawcable.net) |
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| 05:55.10 | *** join/#asterisk jetaway2009 (n=asd@218.111.223.133) |
| 05:56.15 | jetaway2009 | hi |
| 05:56.25 | jetaway2009 | anyone here? |
| 05:56.44 | jetaway2009 | all sleeping ? |
| 05:59.29 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 06:03.05 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001217014b92.vc.shawcable.net) |
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| 06:20.36 | mitcheloc | file: i just tried gnutls-devel and it installed, same problem, i'll recompile asterisk as well maybe it has to be done after the package is installed? |
| 06:22.06 | hads|home | most likely |
| 06:23.26 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 06:23.53 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:34.59 | mitcheloc | nope, that didn't help it... still gettingthe error asking for it |
| 06:37.40 | *** join/#asterisk NoRemorse (n=bah@210-84-10-183.dyn.iinet.net.au) |
| 06:37.44 | *** join/#asterisk tparcina (n=tparcina@lns01-0238.dsl.iskon.hr) |
| 06:37.47 | NoRemorse | hi all |
| 06:37.48 | hads|home | :/ I know nothing of jingle |
| 06:38.08 | NoRemorse | I an getting a seg fault in 1.2.10 when it starts up cdr_mysql are there any known issues? |
| 06:38.48 | mitcheloc | hads|home, it looks so enticing and i can't use it! *ugh* |
| 06:40.20 | *** join/#asterisk pardove (n=pardove@217.219.250.25) |
| 06:40.52 | pardove | how can i join two * servers using SIP? |
| 06:41.34 | NoRemorse | sip peer |
| 06:46.30 | pardove | anybody in? |
| 06:46.32 | pardove | how can i join two * servers using SIP? |
| 06:48.08 | benjk | anybody here using Ubuntu? |
| 06:50.05 | NoRemorse | sip peer... i said it already |
| 06:50.21 | *** join/#asterisk Dieno (n=CoOLmiRC@58.65.197.19) |
| 06:50.28 | Dieno | hello |
| 06:50.37 | *** join/#asterisk Azrael (n=Azrael@orion.negativeblue.com) |
| 06:50.37 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 06:51.00 | Dieno | can n e help me abt Fedora 4 Installation |
| 06:51.10 | Assid | man my polycom301 time keeps slowing down... |
| 06:53.06 | Dieno | can n e 1 tell me which type of RED Hat Require like server, workstation or samba server for asterisk |
| 06:54.31 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 06:54.46 | pdtmobile | Dieno: pick whichever one you want, you can add and remove packages if you need to after you install |
| 06:55.05 | Dieno | ohke |
| 06:55.11 | pdtmobile | i would suggest server, it doesn't have all the junk, it has a bunch of junk but not all of it :) |
| 06:55.14 | Dieno | thnx |
| 06:55.20 | JT | Dieno is an onjoin spammer |
| 06:55.35 | Dieno | wat |
| 06:55.55 | JT | as soon as i joined the chan |
| 06:55.56 | JT | i get |
| 06:55.57 | *** join/#asterisk cybergypsy (n=mark@APoitiers-157-1-36-162.w83-193.abo.wanadoo.fr) |
| 06:55.57 | JT | 16:51 -!- Irssi: Starting query in freenode with Dieno |
| 06:55.58 | JT | 16:51 <Dieno> can u tell me which type of RED Hat Require like server, |
| 06:55.58 | JT | <PROTECTED> |
| 06:56.00 | JT | 16:51 <Dieno> lpz |
| 06:56.03 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 06:56.17 | Dieno | mis typin |
| 06:56.40 | JT | you mistyped all that stuff and accidentally sent it to me? |
| 06:56.58 | Dieno | not accidently |
| 06:57.09 | Dieno | i want to know dats y i sent it to u |
| 06:57.33 | Dieno | no bdz answering dats y i am asking every 1 seperatly |
| 06:57.41 | Dieno | now i got my prb solve |
| 06:57.56 | Dieno | so plz wat ever u wan to do wid my nik or ip |
| 06:58.00 | Dieno | u can |
| 06:58.00 | Dieno | :D |
| 06:58.27 | JT | yeah anyway |
| 06:58.57 | JT | it's considered poor netiquette to peristantly engage in uninvited questioning of people on a channel |
| 06:59.02 | JT | via private message |
| 06:59.47 | type0 | haha |
| 06:59.52 | Dieno | oh its hurt |
| 06:59.54 | Dieno | i am goin |
| 06:59.55 | Dieno | :D |
| 07:00.03 | JT | and people are possibly also ignoring you due to poor english |
| 07:00.13 | JT | unnecessary abbreviations |
| 07:02.05 | Assid | Dieno: do not pm |
| 07:05.45 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 07:08.48 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B6AEE6.pool.t-online.hu) |
| 07:09.06 | *** join/#asterisk Shursander (n=Shursand@213.137.247.78) |
| 07:09.11 | Shursander | hello |
| 07:09.28 | *** join/#asterisk schurzi (n=schurzi@www.verdammte-seelen.de) |
| 07:11.15 | Shursander | plz help me! As to make so as to would work in astkrisk it would work the function CHANNEL? |
| 07:11.52 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 07:12.27 | Shursander | Asterisk 1.2.9.1 |
| 07:14.10 | pardove | NoRemorse: explain more plz |
| 07:15.07 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.162) |
| 07:16.09 | pardove | how can i join two * servers using SIP? |
| 07:17.21 | kmilitzer | pardove: Just direkt the SIP messages from one to another. ;) Give more details, I don't understand exactely what you want. |
| 07:17.59 | *** part/#asterisk WiredX (n=OzPacifi@home.ozpacific.net.au) |
| 07:22.32 | MrChimpy | morning |
| 07:24.22 | *** join/#asterisk vgster (n=vgster@217.78.147.238) |
| 07:25.09 | pardove | kmilitzer: i want extens on ServerA call extens on ServerB |
| 07:25.35 | Assid | pardove: look into iax links or sip links.. |
| 07:25.38 | Assid | trunking |
| 07:25.52 | *** join/#asterisk cvv (n=cvv@212.8.35.34) |
| 07:25.53 | pardove | kmilitzer: all using SIP, dont want to use IAX |
| 07:26.16 | cvv | good morning! |
| 07:27.35 | kmilitzer | pardove: Do the Servers have to authenticate, or are they trustworthy? |
| 07:27.41 | JT | pardove: why? |
| 07:27.55 | kmilitzer | Folks, IAX is not always the way to go ... |
| 07:28.08 | cvv | please recommend a good stable bugless SIP client for Linux and for windows |
| 07:28.43 | kmilitzer | ... it makes you depenent on asterisk. With SIP you can change components in your VoIP-System regardless if it's an asterisk or a CCM |
| 07:29.12 | MrChimpy | cvv: anything else? would you like that in gold, sir? |
| 07:29.17 | tparcina | easy question, buz asking just to be sure - i'm logging CDR data to MYSQL database using ODBC driver. I have created database as usaly, with all neaded feilds. now, because of easier manipulation with data I need to put one extra field in database. that will be unique incremented number. will that new field make problems for loging asterisk CDR data to that MYSQl database? as far as I (with my minor knowledge of database) know it shouldn't ma |
| 07:29.43 | tparcina | make any diference. now, i just would like someone to condirm or deny this. thank you! |
| 07:30.43 | tparcina | cvv: don't use SJPhone, I head serious problems. now i have uninstall them and bouth hardphone |
| 07:30.54 | cvv | at now i'm tried to use sjphone and x-lite. I found out then x-lite is buggy and sjphone is very restricted |
| 07:32.43 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 07:32.43 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:32.59 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:33.57 | pardove | JT: i want to have video calls as well so IAX can't help |
| 07:34.31 | *** join/#asterisk Gr1ncheux_ (n=devine@AStDenis-105-1-39-190.w80-8.abo.wanadoo.fr) |
| 07:34.44 | JT | ok |
| 07:35.53 | pardove | JT: i can't find any clear soloution for sip to sip call between two * servers, any exprience here? |
| 07:37.52 | pardove | kmilitzer: about the server authentication, it doesn't make any sense |
| 07:38.19 | kmilitzer | pardove: It's easy. Define a peer in sip.conf with host=IP of the other host for both systems ... |
| 07:38.46 | pardove | kmilitzer: does it work with user=friend ? |
| 07:39.02 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 07:39.03 | kmilitzer | pardove: Then just do a Dial(SIP/exten@<peername>) |
| 07:39.13 | kmilitzer | pardove: I have type=peer |
| 07:39.29 | pardove | kmilitzer: thx, let me test |
| 07:43.41 | pardove | kmilitzer: it says, SIP/192.168.2.1-081858b0 is circuit-busy |
| 07:44.10 | pardove | kmilitzer: my dial cmd is: Dial(SIP/12@192.168.2.1,30,r) |
| 07:44.35 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 07:45.08 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 07:46.19 | pardove | kmilitzer: also tested it by: Dial(SIP/12@100,30,r) which 100 is the peername. the same result :( |
| 07:48.34 | E-bola | How can i check that a host has a sip proxy running? |
| 07:48.47 | E-bola | is there some tool to monitor if a sip proxy is up? Remotely that is |
| 07:48.49 | pardove | kmilitzer: now it says-> SIP/100-08182c28 is circuit-busy |
| 07:49.04 | mitcheloc | um did anyone else just get an e-mail from an alex huang? |
| 07:49.25 | mitcheloc | it looks like he spammed the entire mailing list... |
| 07:51.33 | kmilitzer | pardove: Can you make a SIP trace with ngrep? |
| 07:52.44 | pardove | kmilitzer: on which box, caller or callee? |
| 07:52.47 | kmilitzer | E-bola: qualify=yes for a peer!?! |
| 07:52.53 | X-Gen | someone has rev-eng skype :) |
| 07:53.06 | kmilitzer | pardove: doesn't matter I think, but start on the caller side |
| 07:53.13 | *** join/#asterisk cybergyp1y (n=mark@90.5.58.30) |
| 07:53.27 | kmilitzer | Maybe you can make asterisk more verbose at first |
| 07:53.30 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 07:53.30 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:53.34 | *** join/#asterisk |oranjia| (n=kvirc@dsl-146-34-237.telkomadsl.co.za) |
| 07:53.36 | kmilitzer | set verbose 5 for example on the CLI |
| 07:53.45 | |oranjia| | hello peeps :) |
| 07:53.53 | E-bola | kmilitzer: im looking for something none asterisk related |
| 07:53.59 | E-bola | asterisks just times out authenticating |
| 07:55.25 | kmilitzer | E-bola: I use sipsak to check, if I can login in my registrar server, but you can do nearly anything with it and it works with nagios |
| 07:55.40 | E-bola | i tried the nagios plugin |
| 07:55.45 | E-bola | ile try sipsak |
| 07:55.51 | pardove | kmilitzer: does a sip debug help instead ngrep, i dont have ngrep on my box ;-) |
| 07:57.25 | kmilitzer | pardove: I guess so. I always use tethereal, but sip debug should do the same |
| 07:57.59 | |oranjia| | can anyone help me understand why i'm getting this message : Connect attempt from '127.0.0.1' unable to authenticate |
| 07:58.30 | |oranjia| | i've been checking and rechecking the manager.conf files...and i can't seem to see where the problem is |
| 07:58.41 | *** join/#asterisk c1sco (n=loke@c-24-5-215-69.hsd1.ca.comcast.net) |
| 07:58.43 | c1sco | hello |
| 07:58.48 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 07:58.52 | E-bola | kmilitzer: can u show me how i woudl test with sipsak? |
| 07:58.54 | E-bola | i just compiled it |
| 07:59.06 | c1sco | i got asterisk up and running, but i am having issues with moh |
| 07:59.19 | |oranjia| | http://pastebin.com/750041 |
| 07:59.26 | kmilitzer | E-bola: man sipsak ;) |
| 07:59.35 | c1sco | i ran lame from the addons packages and edited the musiconhold.conf but there is just silence |
| 08:00.00 | E-bola | kmilitzer: :( |
| 08:00.20 | kmilitzer | E-bola: What else should I tell you? I don't know what exactley you want to test ... |
| 08:00.30 | c1sco | i am debuging sip and see started moh, then stopped moh |
| 08:01.04 | kmilitzer | E-bola: sipsak has a good man page, there you should find anything you need to know |
| 08:01.50 | E-bola | well i guess im not sure what i need to know hehe |
| 08:01.58 | E-bola | there dont seam to be a diagnostic command |
| 08:02.00 | E-bola | or anything similar |
| 08:03.30 | c1sco | any ideas on how to get my moh working? |
| 08:06.05 | E-bola | ahh i do get something back |
| 08:06.10 | E-bola | why does asterisk timeout then |
| 08:17.38 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:19.10 | JT | anyone built asterisk as a least cost router or a pabx gateway? |
| 08:24.49 | *** join/#asterisk kristalino (n=kristali@84-50-84-146-dsl.trt.estpak.ee) |
| 08:26.58 | *** join/#asterisk saftsack (n=saftsack@p54A7F2AB.dip.t-dialin.net) |
| 08:29.11 | *** join/#asterisk tparcina (n=tparcina@lns01-0238.dsl.iskon.hr) |
| 08:29.32 | e-ddie | does anyone know anything about why i get the sounds in english, when i answer the queue, and then i.e. transfers a call. when i call both via sip and iax from internal and external nr's, and transfer the call, i get it in danish, which is how it's suppposed to be |
| 08:32.50 | tparcina | Asterisk database, where can i find information about maximum data I can store in it? |
| 08:33.42 | tparcina | I'm planing to put almost 1000 callerID records in * internal DB, will it handle it? |
| 08:35.02 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 08:35.02 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:35.15 | saftsack | you mean for screening their names on the displays on the telephones someone phones you? |
| 08:36.11 | *** join/#asterisk jetaway2009 (n=asd@60.50.30.207) |
| 08:36.51 | jetaway2009 | any experienced with channel bank? |
| 08:37.12 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 08:38.55 | jetaway2009 | hello all |
| 08:39.45 | *** join/#asterisk kay2 (n=key2@sd-420.dedibox.fr) |
| 08:43.53 | *** join/#asterisk Mr-packet (n=a@222-154-239-122.adsl.xtra.co.nz) |
| 08:44.32 | *** join/#asterisk ghenry (n=ghenry@80.229.93.1.plusnet.pte-ag2.dyn.plus.net) |
| 08:46.07 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 08:46.10 | Chris-NB | hi |
| 08:46.15 | E-bola | can soembody explain to me why sometimes i cant use exit or quit to quit the asterisk console? |
| 08:46.25 | Chris-NB | I've two asterisk boxes and want to connect them via iax |
| 08:46.29 | E-bola | do it have to be started as a daemon somehow? |
| 08:46.31 | MrChimpy | cos you haven't read any manual? |
| 08:46.33 | Mr-packet | im looking for some pointers on how to set up some Linksys SPA941's with Asterisk.. I'm obvioulsy not googling the right words.. any clues |
| 08:46.33 | Chris-NB | do I have to register each at the other one ? |
| 08:46.56 | MrChimpy | holy crap it's #dumbquestions |
| 08:46.58 | Chris-NB | to be able to call from box1 to box2 and back |
| 08:47.34 | MrChimpy | cnb: look in the sample configs or docs before asking. it's all there. |
| 08:49.03 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.155) |
| 08:49.09 | E-bola | there arent any dumb questions |
| 08:49.13 | E-bola | only dumb idiots claiming so |
| 08:49.33 | Strom_C | what is 2 + 2? |
| 08:49.35 | Strom_C | i dont know |
| 08:49.37 | Strom_C | help plz |
| 08:49.47 | kmilitzer | E-bola: If you start asterisk as daemon and connect with asterisk -r you can use "quit". If you run it in foreground, only a stop <now|when convenient> works |
| 08:50.04 | kmilitzer | E-bola: And asterisk tells you that!!!!!!! |
| 08:50.05 | E-bola | kmilitzer: thanks, then it is as i suspected |
| 08:51.00 | MrChimpy | e-bola: there are dumb questions when the answers are clearly documented and you're just being lazy. |
| 08:51.08 | kmilitzer | Strom_C: I think it's 5, but I am not sure |
| 08:51.18 | E-bola | mrchimpu: thats nonesence |
| 08:51.34 | E-bola | millions of topics are documented, does that mean the students cant ask the teacher questions? |
| 08:52.00 | Strom_C | kmilitzer: but but but voip-info said it was somewhere between six and forty-nine |
| 08:52.10 | MrChimpy | so do did you call the manufacturer to work out how to use your toaster because you couldn't be bothered reading the instructions? |
| 08:52.44 | Strom_C | MrChimpy: you forgot about #toaster |
| 08:52.58 | jetaway2009 | forgive him ppl |
| 08:53.09 | jetaway2009 | ppl do make mistake |
| 08:53.18 | jetaway2009 | pls forget and forgive |
| 08:53.49 | E-bola | u guys need to learn the rtfm expression |
| 08:54.03 | MrChimpy | quite aware of it thanks |
| 08:54.16 | MrChimpy | and if you're aware of it, wtf are you asking us for? |
| 08:55.32 | Mr-packet | not a very friendly bunch here |
| 08:56.27 | jetaway2009 | ppl of the great asterisk channel ...calm urself |
| 08:56.40 | kmilitzer | Mr-packet: The thing is, that people are too lazy to try something for themselves first. Instead they questions that have been answered a hunderd times already |
| 08:56.45 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 08:56.53 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:57.14 | kmilitzer | s/they questions/they ask questions/ |
| 08:57.30 | MrChimpy | yep, people here are very helpful and friendly if you've at least made an effort to solve your own problem, or you're asking stuff that isn't clear. |
| 08:58.05 | E-bola | I think ppl are friendly too |
| 08:58.10 | E-bola | and im used to irc, and yes im lazy |
| 08:58.14 | Mr-packet | jbot: yhes that is quite true, not enough trying |
| 08:58.16 | jbot | okay, Mr-packet |
| 08:58.18 | E-bola | and yes i hate it when users dont try stuff thme selfs |
| 08:58.20 | jetaway2009 | yes..true...ppl of the asterisk are busy ppl |
| 08:58.20 | E-bola | but thats life :) |
| 08:58.22 | Mr-packet | does 2 weeks trying count? |
| 08:58.59 | kmilitzer | Mr-packet: The question is, what you tried for two weeks ;) |
| 08:59.05 | |oranjia| | Connect attempt from '127.0.0.1' unable to authenticate |
| 08:59.12 | MrChimpy | interesting "i hate it when people are idiots. I'm behaving like one. Sorry, that's life" attitude there. |
| 08:59.14 | |oranjia| | can someone please give me some ideas - this error message is killing me. :: |
| 08:59.29 | Mr-packet | i've been trying to get some linksys spa941 phones working with asterisk |
| 08:59.41 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 08:59.41 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:59.58 | MrChimpy | mr-packet: if you've started to bleed with frustration you might be justified in asking :) |
| 09:00.52 | Mr-packet | what fustrated me more was the crappy linksys SPA9000 pabx thing that they sell as well. its just dumb.. so i thought that there was a better way, and found asterisk |
| 09:02.10 | Mr-packet | really a pointer to a good HOWTO on this would be good |
| 09:02.10 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:07.02 | *** join/#asterisk einsiedler (n=einsiedl@p5084AE4E.dip0.t-ipconnect.de) |
| 09:07.05 | kmilitzer | Mr-packet: www.voip-info.org www.astridocs.org (i think that's where the asterisk book is, not sure though) |
| 09:08.29 | hads|home | ~docs |
| 09:08.31 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 09:09.37 | *** join/#asterisk my007ms (n=noor@217.139.224.194) |
| 09:10.01 | my007ms | what make that will make initd script |
| 09:10.08 | my007ms | is it make config |
| 09:10.18 | jetaway2009 | ??!! |
| 09:10.54 | hads|home | erm. |
| 09:10.55 | my007ms | i need make initd script to make asterisk start after reboot |
| 09:11.06 | stoffell_h | anyone already compiled latest bristuff ? (0.3.0-PRE-1r) |
| 09:11.27 | kmilitzer | my007ms: Look in the contrib directory of the sources |
| 09:11.29 | hads|home | my007ms: contrib/init.d/ in your asterisk sauce |
| 09:11.39 | hads|home | Yeah, what he said :) |
| 09:11.40 | *** join/#asterisk alib80 (n=chatzill@196.35.242.16) |
| 09:12.14 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:12.35 | E-bola | My sip->pstn provider gave me an asterisk macro to loadbalance between his 2 sip proxies. Where can i find info on using macros to dial out? |
| 09:13.01 | |oranjia| | has anyone been able to enable fax to email based on extensions and not just Zap channels? |
| 09:13.20 | alib80 | hi all, i was wondering if anyone knew how to ensure that when i drop a call from a sip phone asterisk does not keep dialing the number. Invariably the called person is put on hold. |
| 09:13.21 | kmilitzer | E-bola: Cool, can you pastebin that macro? |
| 09:13.24 | stoffell_h | |oranjia|: you could have a look at iaxmodem wich uses hylafax.. |
| 09:13.51 | |oranjia| | thanks stoffel_h :) |
| 09:14.07 | kmilitzer | E-bola: On the CLI type: show application Macro |
| 09:15.06 | E-bola | kmilitzer: sure just a sec |
| 09:15.31 | E-bola | but wher ecna i find information on how to use macro's? |
| 09:15.39 | kmilitzer | ~docs |
| 09:15.40 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 09:16.01 | kmilitzer | E-bola: Make your choice where to look ;) |
| 09:16.10 | stoffell_h | jetaway2009: u here?:) |
| 09:16.50 | einsiedler | anyone compiled chan_misdn latest trunk ? after starting asterisk and misdn i get undefinded symbol : misdn_lib_l1watcher |
| 09:17.03 | jetaway2009 | yeap |
| 09:17.12 | jetaway2009 | and hiding here |
| 09:18.03 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 09:18.30 | E-bola | kmilitzer: http://pastebin.ca/90078 there u go |
| 09:18.36 | E-bola | its quite simple to comprehend |
| 09:18.59 | E-bola | i can read it pretty acurate, but since im new to asterisk i dont knwo how to set it up |
| 09:19.53 | Mr-packet | does every ip phone that i want to attach to my my asterisk box, need an entry in extenstions.conf? |
| 09:21.15 | *** join/#asterisk parag_ast (n=root@dxb-b111365.alshamil.net.ae) |
| 09:21.45 | parag_ast | can anybody suggest any IAX SDK client to which i can modify according to my need ... |
| 09:26.42 | *** join/#asterisk effectiveape (n=nick@82.153.22.16) |
| 09:26.50 | Chris-NB | what do I have to do, that I can register an asterisk via IAX an another asterisk? |
| 09:27.00 | Chris-NB | or, can someone tell me where to look at? |
| 09:27.45 | effectiveape | Any bri (specifically quadbri) experts out there. |
| 09:28.46 | stoffell_h | effectiveape: you won't know untill you ask :) |
| 09:29.17 | effectiveape | I've got the junghanns stuff all setup (1.2) and am just having some probs with getting it all working :) The card is coming up and everything and connected to the isdn but when i call in i'm getting these probs.... |
| 09:29.27 | effectiveape | (just waiting for pastebin :/) |
| 09:29.35 | parag_ast | can anybody suggest any IAX SDK client to which i can modify according to my need ... |
| 09:30.12 | MrChimpy | my fastagi! it lives! |
| 09:31.40 | effectiveape | this is it if anyone else can get it to come up :/ http://pastebin.com/749829 |
| 09:32.15 | effectiveape | getting -- chan_zap.c:8503 zt_pri_error: 3 !! Got reject for frame 3, retransmitting frame 3 now, updating n_r! |
| 09:33.42 | hads|home | effectiveape: pastebin.ca is faster at times if you are having trouble. |
| 09:35.54 | effectiveape | cool thanks for that.... http://pastebin.ca/90090 |
| 09:36.07 | effectiveape | That is the full output with bri debug on |
| 09:36.28 | effectiveape | i've not setup isdn asterisk before btw ;) |
| 09:37.51 | effectiveape | it's bt isdn2e too if that makes a difference |
| 09:38.44 | stoffell_h | effectiveape: can you paste your zapata.conf also? |
| 09:38.51 | effectiveape | yep..... |
| 09:38.57 | E-bola | Im a bit lost regarding my outgoing macro, do anybody know what to put in exstensions.conf to get all outgoing calls to go through the macro? |
| 09:39.13 | stoffell_h | effectiveape: and the version of bristuff? |
| 09:40.20 | effectiveape | http://pastebin.ca/90093 |
| 09:40.44 | effectiveape | 0.3.0-PRE-1r |
| 09:41.27 | effectiveape | no sorry 0.3.5 (* STABLE) |
| 09:41.34 | *** join/#asterisk faberk64 (n=faberk@213.199.15.249) |
| 09:42.17 | effectiveape | arg no sorry it was 3.0-PRE |
| 09:42.24 | stoffell_h | effectiveape: u mean 0.3.0-pre-1r ? right? (is released today, this morning) |
| 09:42.31 | effectiveape | i think i might have installed the wrong one? |
| 09:42.53 | effectiveape | heh. Right - bristuff-0.3.0-PRE-1q |
| 09:43.06 | stoffell_h | q is okay, but r might be better (if it compiles, here it doesn't) |
| 09:43.26 | effectiveape | should i be on 0.3.5? |
| 09:43.36 | stoffell_h | effectiveape: your zapata.conf looks good.. might try signalling = bri_cpe instaed of bri_cpe_ptmp maybe |
| 09:43.39 | effectiveape | oh no sorry thats chan_capi |
| 09:43.49 | *** join/#asterisk pjo (n=pjo@212.88.98.114) |
| 09:43.59 | stoffell_h | effectiveape: no, your bristuff is okay (but you sould upgrade to pre-1r though..) |
| 09:44.50 | effectiveape | fg |
| 09:44.56 | effectiveape | doh wrong window ;) |
| 09:45.27 | effectiveape | i'm not sure thats the problem though is it? |
| 09:45.42 | effectiveape | i've tried bri_cpe - no diff |
| 09:46.13 | stoffell_h | effectiveape: okay, upgrade to -1r then, there are some notices in the Changelog on libpri/zap detection |
| 09:46.14 | effectiveape | I'm right to have it setup as TE aren't i? |
| 09:46.22 | effectiveape | aha cool. |
| 09:46.32 | stoffell_h | effectiveape: if you connect to the telco, you're right.. (default card behaviour that is) |
| 09:47.01 | effectiveape | if i do the upgrade it;'s not going to scupper my config is it? |
| 09:47.18 | stoffell_h | effectiveape: no, it'll leave your config.. (or backup with tar :)) |
| 09:47.28 | effectiveape | yeah of course ;) |
| 09:48.53 | stoffell_h | effectiveape: zapata.conf looks good, only, dont use # for comments, but ; .. (I also forget that EVERY time :p) |
| 09:49.26 | effectiveape | aha - good call ;) ta. |
| 09:49.40 | effectiveape | Ok downloading updates - so we'll see if that helps. :) |
| 09:49.58 | stoffell_h | effectiveape: ok, let me know if it compiles... i can't get it to .. |
| 09:50.21 | effectiveape | cool - willdo |
| 09:57.52 | tparcina | saftsack> yes, i mean this - you mean for screening their names on the displays on the telephones someone phones you? |
| 10:01.25 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 10:01.25 | *** mode/#asterisk [+o denon] by ChanServ |
| 10:06.52 | _4d4m_ | hi all.. anyone able to save me some googling and tell me if a nokia N91 will work with * using SIP? |
| 10:07.01 | _4d4m_ | got a client asking me |
| 10:07.37 | effectiveape | _4d4m_ www.sillyant.com |
| 10:07.47 | effectiveape | with with n90 so i'm guessing maybe. |
| 10:08.10 | effectiveape | However the N80 doesn't because of nokia's auth for deelopers |
| 10:08.25 | effectiveape | developers even ;) |
| 10:08.59 | effectiveape | stoffell_h: where does your compile fail? |
| 10:13.43 | _4d4m_ | effectiveape: cheers.. much appreciated |
| 10:14.06 | effectiveape | stoffell_h: compiled without problems |
| 10:14.45 | effectiveape | np |
| 10:18.13 | Strom_C | wheeeeeeeeeeeeeee |
| 10:21.27 | tparcina | where can i find which charset's does voicemail.conf support* |
| 10:21.36 | Strom_C | tparcina: ?? |
| 10:21.40 | my007ms | can i use , with channel=> like channel=>1,3,5 |
| 10:22.04 | Strom_C | my007ms: in zapata.conf? |
| 10:22.09 | my007ms | yes |
| 10:22.30 | tparcina | Strom_C: in voicemail.conf there si charset option, what charset's does it support? |
| 10:22.43 | Strom_C | my007ms: not sure whether that's valid syntax, but if you put those on separate lines immediately after one another you should be fine |
| 10:23.12 | Strom_C | my007ms: alternatively, configure your zaptel channels so they're arranged by function and you dont have to worry about noncontiguous blocks :) |
| 10:23.22 | my007ms | u mean channel=>1 |
| 10:23.22 | Strom_C | tparcina: beats me |
| 10:23.26 | my007ms | then channel=>3 |
| 10:23.31 | my007ms | ..etc |
| 10:23.45 | Strom_C | yep |
| 10:23.55 | *** join/#asterisk LakeSolon (n=blake@12-227-169-99.client.mchsi.com) |
| 10:33.28 | *** join/#asterisk _problem_ (n=lokesh_k@estrela.nortenet.pt) |
| 10:37.40 | effectiveape | argh!!!! Now i'm getting Channel 0/1, span 4 got hangup, cause 81 |
| 10:40.42 | *** join/#asterisk breakdisk (n=breakdis@62.149.122.2) |
| 10:40.49 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 10:41.19 | effectiveape | anyone know what 'invalid call reference value' is ? |
| 10:41.28 | effectiveape | which is apparently cause 81 |
| 10:41.45 | breakdisk | hi guys i have a question.i need to know how to implement in a2billing to dial another number. |
| 10:42.40 | breakdisk | in some pbx it says you dial asterisk(*) twice and you can dial another number. |
| 10:52.49 | E-bola | hi guys can u tell me if this pattern (exten => ZZZZZZZZ) will get matched if i dial 51926215 on my sip phone? |
| 10:55.32 | Assid | wouldnt that be XXXXXX ? |
| 10:56.15 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 10:56.56 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 10:57.26 | effectiveape | anyone any insights? ;) |
| 10:59.10 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 10:59.35 | *** part/#asterisk tparcina (n=tparcina@lns01-0238.dsl.iskon.hr) |
| 11:03.08 | effectiveape | still there stoffell_h? |
| 11:04.02 | *** join/#asterisk RoyK[uk] (n=roy@148.122.35.210) |
| 11:05.33 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 11:06.30 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 11:08.20 | stoffell_h | effectiveape: my compile failed at zaptel (tonezone.c) |
| 11:08.31 | stoffell_h | (i was out lunching, just came back ;)) |
| 11:08.39 | RoyK[uk] | morning |
| 11:08.42 | effectiveape | aha cool :) |
| 11:08.54 | stoffell_h | no, it's hot... 33 degr celsius here :) |
| 11:08.59 | *** join/#asterisk ChrisDE4 (n=ChrisDE@tmo-078-86.customers.d1-online.com) |
| 11:09.09 | effectiveape | no problems at all with my compile. |
| 11:09.16 | effectiveape | Where are you? |
| 11:09.29 | stoffell_h | in europe, belgium |
| 11:09.37 | RoyK[uk] | round-trip min/avg/max/stddev = 639.723/2079.820/3426.707/713.420 ms |
| 11:10.04 | stoffell_h | RoyK[uk]: so you're pretty high at the moment? |
| 11:10.04 | effectiveape | dunno if any any further or not but i'm getting different error ;) |
| 11:10.13 | ChrisDE4 | hi. whats this: Jul 15 17:52:51 DEBUG[18440] channel.c: Nobody there, continuing... |
| 11:10.25 | RoyK[uk] | stoffell_h: about five meters above sea level |
| 11:10.30 | ChrisDE4 | if asterisk recognizes that noone is there, why doesn't it hangup? |
| 11:10.48 | *** join/#asterisk VeNoMouS_ (n=jj@nibbler.turnstone.net.nz) |
| 11:11.03 | VeNoMouS_ | hay does anyone in here have eyebeam video working with asterisk |
| 11:11.19 | RoyK[uk] | VeNoMouS_: have tried it, yes |
| 11:11.20 | stoffell_h | RoyK[uk]: lol, you're really surfin' then |
| 11:11.36 | RoyK[uk] | :) |
| 11:11.38 | stoffell_h | effectiveape: paste your error on pastebin? |
| 11:11.38 | RoyK[uk] | ferry |
| 11:11.40 | effectiveape | heh |
| 11:11.42 | stoffell_h | hehe |
| 11:11.42 | VeNoMouS_ | RoyK[uk] : im having issues aye, video isnt working |
| 11:11.43 | RoyK[uk] | newcastle-bergen |
| 11:11.51 | VeNoMouS_ | even tho im doing videosupport=yes and allow=h263 |
| 11:11.52 | RoyK[uk] | ~video |
| 11:11.53 | jbot | methinks video is interrupt 10h for the video BIOS |
| 11:11.57 | VeNoMouS_ | i can see the rtp stream going in |
| 11:12.08 | VeNoMouS_ | but nothing on the other end |
| 11:12.21 | VeNoMouS_ | asterisk opens up a port for the rtp of the video |
| 11:12.28 | VeNoMouS_ | i can see if when i tcpdump |
| 11:12.34 | VeNoMouS_ | but it just dont pass it on |
| 11:13.22 | RoyK[uk] | sorry. don't know. worked last i tried it |
| 11:13.41 | VeNoMouS_ | were you testing on internal lan? |
| 11:13.52 | VeNoMouS_ | or on a remote asterisk? |
| 11:14.07 | effectiveape | stoffell_h: yep.... http://pastebin.ca/90135 |
| 11:19.08 | *** join/#asterisk Dr-Linux|work (n=Nothing@202.125.139.198) |
| 11:21.48 | E-bola | Do anybody know how i can get the stuff i see on the asterix console into a file? |
| 11:24.05 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 11:25.55 | stoffell_h | E-bola: check /var/log/asterisk/full |
| 11:25.59 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 11:26.22 | stoffell_h | effectiveape: ouch, that doesn't look like a good error.. |
| 11:26.28 | E-bola | stoffell: it doesnt exist |
| 11:26.34 | effectiveape | i'm thinking it's something to do that that cause 81 but my googleing has yielded no answer |
| 11:26.41 | stoffell_h | E-bola: then logger.conf is your friend (in /etc/asterisk) |
| 11:26.48 | E-bola | ok lemme read that |
| 11:26.59 | stoffell_h | effectiveape: and changing the bri_cpe mode? |
| 11:27.17 | stoffell_h | it's safe to try all others, it just won't work if it's not correct :) |
| 11:27.25 | effectiveape | when i change it to that it goes mental ;) |
| 11:28.26 | VeNoMouS_ | grrr |
| 11:28.27 | VeNoMouS_ | ffs |
| 11:28.35 | VeNoMouS_ | i can see the video stream hitting asterisk |
| 11:28.39 | VeNoMouS_ | and it not going past that |
| 11:28.45 | effectiveape | ooh hang on |
| 11:28.49 | VeNoMouS_ | it even shows in the rtp debug |
| 11:29.21 | effectiveape | it's looping this.... http://pastebin.ca/90143 |
| 11:29.31 | effectiveape | But i have contact! ;) |
| 11:29.38 | effectiveape | It's well distorted though |
| 11:30.15 | *** part/#asterisk pjo (n=pjo@212.88.98.114) |
| 11:30.56 | stoffell_h | effectiveape: canyou show the zapata.conf again? |
| 11:31.01 | stoffell_h | the new one |
| 11:32.00 | effectiveape | yep.... http://pastebin.ca/90147 |
| 11:32.10 | znoG | hey, if I want to limit how long a call can go for... I tried using the L parameter for Dial which works mostly but it has failed a couple of times to terminate a call. Should I use AbsoluteTimeout? |
| 11:33.24 | stoffell_h | effectiveape: you're getting closer it seems.. you're using connections 1 and 2 ? or are all ports on the quadbri filled? |
| 11:33.37 | effectiveape | aha no. Only 2 ports |
| 11:33.58 | effectiveape | Couldn't see a number on them they are the leftmost 2 |
| 11:34.11 | effectiveape | i'm guessing i need to deactivate the rest |
| 11:34.37 | stoffell_h | effectiveape: you can: http://pastebin.ca/90148 |
| 11:35.05 | *** join/#asterisk kindor (n=roy@office.open-ict.nl) |
| 11:35.22 | kindor | hello |
| 11:37.02 | effectiveape | fantastic! |
| 11:37.15 | effectiveape | Just need to sort the distortion now ;) |
| 11:37.55 | *** join/#asterisk NoRemorse (n=bah@210-84-10-183.dyn.iinet.net.au) |
| 11:37.57 | NoRemorse | hi all |
| 11:38.03 | effectiveape | hey |
| 11:38.15 | NoRemorse | does anyone know if asterisk can talk to an avaya 8300? |
| 11:38.49 | dpryo | It can via pri ;P |
| 11:39.17 | NoRemorse | has to be h.323 is that a prob? |
| 11:39.38 | dpryo | Good luck. |
| 11:39.40 | stoffell_h | effectiveape: cool |
| 11:39.45 | dpryo | I've actually seen somebody do it. |
| 11:39.52 | dpryo | But I still don't believe it. |
| 11:39.56 | NoRemorse | lol |
| 11:40.06 | dpryo | My avaya is connected through t1. |
| 11:40.07 | NoRemorse | why? asterisk does h.323? |
| 11:40.14 | dpryo | Yes, it does. |
| 11:40.31 | NoRemorse | ok cool, I know that cisco claim their are h.323 integration issues with avaya thats all |
| 11:41.29 | effectiveape | stoffell_h: know why it might be doing that? |
| 11:42.05 | VeNoMouS_ | http://pastebin.ca/90150 even with this striped down config video on asterisk isnt going, anyone have any ideas? and yes the box is fully open no firewall rules blocking it, i can see the sender of the video comming in the rtp hits asterisk but asterisk does not on send it to the other client |
| 11:42.11 | stoffell_h | effectiveape: are you sure you're using the correct 2 connections? (maybe try the other 2) |
| 11:42.40 | effectiveape | I'm actually on 3 and 4 - but i've deactivated 1 and 2 |
| 11:42.55 | effectiveape | Thats stopped the erros |
| 11:43.00 | effectiveape | errors even |
| 11:43.18 | stoffell_h | effectiveape: okay, and now you have any other issues ? |
| 11:43.53 | effectiveape | i've got asterisk playing back the trasnfer|hold message but its really distorted |
| 11:44.38 | stoffell_h | effectiveape: that might have many reasons.. cpu usage too high, irq sharing, wrongly formatted audio files... what phones are you using? |
| 11:45.12 | effectiveape | it's not going to the phones yet. I've just called in on my mobile to the number the isdn is on. |
| 11:45.14 | einsiedler | anybody here using sipgate.de ? have some problem while registering after reconnect the dsl line |
| 11:45.15 | NoRemorse | would 2 asterisk boxes be able to be configured to behave like a point to pojnt E1? ie E! over IP? |
| 11:45.34 | effectiveape | And asterisk is playing back one of it's default messages |
| 11:46.35 | stoffell_h | effectiveape: hm, what codec are you using? (alaw/ulaw?) |
| 11:47.34 | VeNoMouS_ | NoRemorse : thats what iax is for |
| 11:47.35 | *** join/#asterisk ghenry (n=ghenry@80.229.93.1.plusnet.pte-ag2.dyn.plus.net) |
| 11:47.38 | effectiveape | does that apply on incomming calls? |
| 11:47.59 | stoffell_h | effectiveape: what distro are you using? |
| 11:48.05 | effectiveape | stoffell_h: ubuntu |
| 11:48.17 | stoffell_h | effectiveape: oh, on incoming calls, you should use the... uhm, raw audio files then, then the distortion is gone |
| 11:49.38 | effectiveape | stoffell_h: sorry (as you can tell) i'm a bit of a noob at this ;) I've currently goy Playback(transfer,skip) |
| 11:49.47 | effectiveape | should i change that? |
| 11:52.48 | stoffell_h | effectiveape: no, it's the audio file that's the reason |
| 11:54.19 | effectiveape | okedoke cool. I'll have a mess about with getting it through to a sip phone. |
| 11:54.26 | NoRemorse | no iax is to connect 2 asterisk servers, i would like to see iax conenct 2 siemens switches... |
| 11:54.33 | effectiveape | Thanks for your help though! :) |
| 11:54.45 | NoRemorse | put a * box on either end with an e1 port and we getting closer |
| 11:55.15 | NoRemorse | purely froma telco point of view, rather than sya pay BT for an E1 between countries... |
| 11:55.28 | NoRemorse | anyways.. gotta go see yas |
| 11:56.14 | stoffell_h | effectiveape: ok,i'll try 2 find the link on those audio files and send it through |
| 11:56.29 | effectiveape | excellent, thanks. |
| 11:57.40 | *** join/#asterisk viperdude (n=jon@philspc.enta.net) |
| 12:00.11 | effectiveape | another quick question whist i'm here. When a call comes in it's looking to match the incomming phone number. Should 's' match too or would i need to do any 6 digit phone number? |
| 12:00.27 | effectiveape | (because 's' isn't matching atm ;)) |
| 12:02.09 | effectiveape | i'm just getting Extension '642042' in context 'inbound-from-isdn' from '' does not exist. |
| 12:06.14 | stoffell_h | effectiveape: check thse 2 links: http://astrecipes.net/?n=152 and http://astrecipes.net/index.php?from=152&q=astrecipes/script+to+convert+music-on-hold+to+native+formats |
| 12:06.51 | stoffell_h | effectiveape: what you want to do is "avoid transcoding", it makes the sounds *much* better when dialing into the isdn |
| 12:08.57 | effectiveape | aha yeah that makes sense. I'll give it a shot, thanks again. |
| 12:18.37 | *** join/#asterisk RoyK (n=roy@148.122.35.210) |
| 12:18.45 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 12:22.46 | stoffell_h | effectiveape: are you using gcc v4 ? (gcc -v) |
| 12:23.44 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 12:24.22 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
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| 12:33.24 | stoffell_h | anyone with debian sarge (gcc 3.3.15 or so) been able to compile bristuff 0.3.0-pre-1r (latest release) ? |
| 12:33.43 | RoyK | stoffell_h: perhaps you'd better pastebin whatever errors you get |
| 12:35.25 | stoffell_h | RoyK: you are so right.. sorry.. |
| 12:35.53 | stoffell_h | i get problems on compiling zaptel : http://pastebin.ca/90169 |
| 12:37.01 | *** join/#asterisk vlt (n=dm@port-87-234-124-192.dynamic.qsc.de) |
| 12:37.06 | vlt | Hello. |
| 12:38.05 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:39.37 | vlt | I'm running asterisk on ubuntu Dapper. I'm connected to a SIP account on the internet via SDSL and some local phones are connected to asterisk (sip). Where can I find out what IP packets/ports I have to give higher priority in traffic shaping? |
| 12:40.05 | effectiveape | that's a weird failure stoffell |
| 12:40.20 | effectiveape | What distro? |
| 12:40.26 | *** join/#asterisk `Kevin (n=Kevin@64.243.236.20) |
| 12:40.59 | *** join/#asterisk trimi` (i=dade@62.162.242.104) |
| 12:41.04 | vlt | Ubuntu Dapper 6.06 |
| 12:41.38 | RoyK | stoffell_h: hm... |
| 12:41.53 | trimi` | can any1 tell me why do my calls get discoonected when im using IAX?? i get this error |
| 12:41.54 | trimi` | Jul 17 14:00:46 WARNING[19023]: chan_iax2.c:1598 attempt_transmit: Max retries exceeded to host xx.xx.xx.xx on IAX2/it-1 (type = 6, subclass = 12, ts=160006, seqno=52) |
| 12:42.03 | stoffell_h | weird enough, it does work with bristuff -1q without problems.. |
| 12:42.29 | effectiveape | have you tried compilling 1q recently? |
| 12:42.42 | stoffell_h | doing the same on a different machine, with gcc 4, that also works.. (it's a different machine with debian testing, not sarge) |
| 12:43.13 | stoffell_h | effectiveape: not since last install |
| 12:43.14 | stoffell_h | :) |
| 12:43.29 | effectiveape | sure something else hasn't changed ;) |
| 12:43.47 | trimi` | Jul 17 14:00:46 WARNING[19023]: chan_iax2.c:1598 attempt_transmit: Max retries exceeded to host xx.xx.xx.xx on IAX2/it-1 (type = 6, subclass = 12, ts=160006, seqno=52) why do i get this error ???????? |
| 12:43.57 | stoffell_h | effectiveape: not that I know of .. |
| 12:43.58 | stoffell_h | :) |
| 12:44.03 | effectiveape | heh ;) |
| 12:44.11 | tzafrir_laptop | stoffell_h, bristuff 1r uses zaptel 1.2.6?? |
| 12:44.12 | effectiveape | you know what you have to do ;) |
| 12:44.33 | stoffell_h | tzafrir_laptop: yeah |
| 12:45.03 | tzafrir_laptop | stoffell_h, if you really want, I've adjusted the bristuff 1q patch to asterisk 1.2.10 (and to zaptel 1.2.7) |
| 12:45.22 | tzafrir_laptop | already commited to the pkg-voip repository |
| 12:45.25 | stoffell_h | i just found out my cat /proc/version says "gcc 3.3.5-12" and gcc -v says "3.3.5-13" |
| 12:45.29 | stoffell_h | could that solve it? |
| 12:45.47 | RoyK | stoffell_h: http://www.mail-archive.com/debian-bugs-dist@lists.debian.org/msg209603.html |
| 12:45.50 | stoffell_h | tzafrir_laptop: hm, interesting, could be worth trying |
| 12:45.53 | RoyK | stoffell_h: i somehow doubt it |
| 12:46.03 | RoyK | but it might be worth a try |
| 12:46.44 | stoffell_h | RoyK: I have a different machine running gcc 3.3.5-13 and /proc/version also says 3.3.5-13, but same error, ... damn.. |
| 12:46.59 | RoyK | tried 3.4? |
| 12:47.29 | stoffell_h | RoyK: apt-get'ting now.. |
| 12:48.06 | tzafrir_laptop | http://svn.debian.org/wsvn/pkg-voip/asterisk/trunk/debian/patches/bristuff.dpatch?op=file&rev=0&sc=0 |
| 12:48.09 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 12:49.24 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 12:49.58 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:50.00 | stoffell_h | RoyK: tanks a lot.. it seems to compile fine now... trying it out further.. |
| 12:50.39 | RoyK | stoffell_h: see the patch from tzafrir_laptop. it looks like a more updated version of bristuff |
| 12:51.10 | stoffell_h | RoyK: ok |
| 12:51.23 | stoffell_h | tzafrir_laptop: thanks, will look into it.. |
| 12:51.25 | E-bola | http://pastebin.ca/90175 Can somebodsy help with this the content of that pastebin? I got a problem with pattern matching |
| 12:52.24 | *** join/#asterisk ariel_ (n=Ariel@70-46-87-158.ftl.fdn.com) |
| 12:52.26 | E-bola | or do any1 have a place where u can try out pattern matching to quickly test what is matched and what isnt? |
| 12:52.45 | *** join/#asterisk jetaway2009 (n=asd@218.111.223.78) |
| 12:53.03 | jetaway2009 | hi |
| 12:54.17 | *** join/#asterisk ZeNiTHaR (n=ask@host.34.62.23.62.rev.coltfrance.com) |
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| 12:56.30 | [TK]D-Fender | E-bola : What the heck is "z" in a dialstring pattern? |
| 12:57.31 | E-bola | a number between 1 and 9 |
| 12:58.07 | E-bola | i just dont get how it works for one number and not another |
| 12:58.12 | many | XZN |
| 12:58.26 | many | [3-9] [4-9] |
| 12:58.28 | many | ... |
| 12:58.29 | many | :) |
| 12:58.32 | [TK]D-Fender | E-bola : Then why are you dialing a number with a lot of 0's in it? |
| 12:59.10 | E-bola | lol |
| 12:59.15 | E-bola | omg im so stupid |
| 12:59.17 | tzafrir_laptop | RoyK, it's not a "more up-to-date version' It's previous bristuff with minor adjustments to make it build with recent zaptel/asterisk |
| 12:59.17 | tzafrir_laptop | actually I'm not sure I needed to touch the zaptel patch at all |
| 12:59.20 | [TK]D-Fender | (duh) |
| 12:59.24 | tzafrir_laptop | And it's not really tested |
| 12:59.24 | E-bola | i just wanted it to not start with a 0 |
| 12:59.34 | [TK]D-Fender | E-bola : BIG PRINT |
| 12:59.37 | E-bola | hehe |
| 12:59.42 | E-bola | its too hot in here darnit!!! |
| 12:59.48 | E-bola | 28 celcios degrees |
| 12:59.51 | E-bola | is to hot to config shit :( |
| 13:00.01 | stoffell_h | E-bola: same here... you're in europe? :p |
| 13:00.32 | [TK]D-Fender | E-bola : My apt hit the 30's this weekend w/o AC. |
| 13:00.42 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:01.08 | E-bola | stoffell: yep denmark |
| 13:01.16 | E-bola | its horrible and im in the middle of a building |
| 13:01.18 | E-bola | so no airflow either |
| 13:02.53 | stoffell_h | E-bola: hm, same here,in belgium... would be better on a beach.. |
| 13:03.32 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:03.32 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:03.57 | Dr-Linux|work | guys, when i do "show channels" i got few, how can i verify which channel is hanged? and i should kill it/ |
| 13:03.58 | Dr-Linux|work | ? |
| 13:08.01 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:08.01 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:12.01 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 13:12.48 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B79EF9.pool.t-online.hu) |
| 13:20.31 | *** join/#asterisk Skarmeth (n=Skarmeth@201008240231.user.veloxzone.com.br) |
| 13:21.26 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 13:21.43 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 13:24.34 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 13:30.19 | *** join/#asterisk RoyK (n=roy@148.122.35.210) |
| 13:33.47 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-136-97.dslextreme.com) |
| 13:38.43 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 13:47.15 | *** join/#asterisk Dovid (n=dovi5988@ool-44c051ad.dyn.optonline.net) |
| 13:49.16 | E-bola | asterisk cant support vbr mp3's for music on hold? |
| 13:50.27 | Dovid | whats vbr ? |
| 13:50.36 | RoyK | variable bitrate? |
| 13:50.45 | Dovid | cant help u there |
| 13:50.57 | Dovid | i know its picky on if u have id3 tags |
| 13:51.12 | RoyK | E-bola: transcode it to the codec you're using |
| 13:51.18 | RoyK | E-bola: far better, and uses less cpu |
| 13:53.36 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:53.40 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 13:53.42 | cy3o3 | Any reccommendations for a good VoiP provider? |
| 13:53.47 | *** join/#asterisk qdk_ (n=qdk@213.150.62.19) |
| 13:54.00 | jetaway2009 | voipjet |
| 13:54.17 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 13:54.26 | MrChimpy | royk: i was under the impression music on hold only took mp3, but i might be talking crap |
| 13:54.35 | cy3o3 | cool |
| 13:54.43 | cy3o3 | hmm, does voipjet allow cid spoofing? :P |
| 13:54.56 | cy3o3 | Nice.. good prices |
| 13:55.49 | jetaway2009 | yes |
| 13:56.50 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 13:58.11 | MrChimpy | http://bugs.digium.com/view.php?id=7498 (my clid spoof patch for 1.2.9.1) |
| 13:59.45 | pdtmobile | lol, i like the discription "secure" so secure I can spoof and make it think i'm the right guy |
| 13:59.50 | pdtmobile | description even |
| 14:00.24 | MrChimpy | but I am the right guy :) |
| 14:01.06 | MrChimpy | clients of mine "secure" their studio phones that we forward winning callers on to by setting the CLID to a known number |
| 14:01.20 | *** join/#asterisk NLinington (n=nfl@82-69-27-212.dsl.in-addr.zen.co.uk) |
| 14:01.33 | pdtmobile | hehe |
| 14:03.02 | NLinington | Hi all, does anybody know how to change the volume levels on a IAX call without using the phones amplify buttons? I am having problems on some long distance calls where the other end cannot here me yet I can here them fine. |
| 14:04.04 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 14:08.11 | *** join/#asterisk Inkubot (n=inkubot@200.119.231.185) |
| 14:08.12 | Inkubot | hi |
| 14:08.16 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 14:08.18 | Inkubot | greets |
| 14:08.35 | Inkubot | where i can find the asterisk sounds in alaw/ulaw format ? |
| 14:09.30 | stoffell_h | Inkubot: http://astrecipes.net/?n=152 |
| 14:09.41 | Inkubot | thanks stoffell_h |
| 14:11.34 | Dr-Linux|work | guys, when i do "show channels" i got few, how can i verify which channel is hanged? and i should kill it? |
| 14:11.57 | Dr-Linux|work | hhm.. |
| 14:12.08 | Dr-Linux|work | never got a answer for this ... anway .. |
| 14:12.29 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:12.34 | *** part/#asterisk ZeNiTHaR (n=ask@host.34.62.23.62.rev.coltfrance.com) |
| 14:13.16 | stoffell_h | Dr-Linux|work: what do you mean by "hanged" ? |
| 14:14.10 | Dr-Linux|work | stoffell_h: like i got a channel is bridged when i do "show channels" but that user is not using it, so it's mean it's hanged. |
| 14:14.58 | Dr-Linux|work | but i hve no way to verify if the user is talking or it's hanged. so whenever i suspect i call the user and verify and he/she says i'm not talking .. so i soft hanup the channel |
| 14:15.54 | stoffell_h | Dr-Linux|work: you can do a "zap destroy channel <nr>" |
| 14:16.35 | Dr-Linux|work | stoffell_h: well, i know that, but how can i make sure, if a call is hanged or user is still talking/using it? |
| 14:18.41 | pdtmobile | Dr-Linux|work: you tried this http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy ? |
| 14:19.03 | stoffell_h | Dr-Linux|work: hm, oke, good question. zap show channel and compare at the moment you've got a hanging and a normal channel.. ?? |
| 14:19.17 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 14:19.33 | Dr-Linux|work | not zap show channels |
| 14:19.41 | Dr-Linux|work | it should be then "sip show channels" |
| 14:20.05 | *** join/#asterisk RoyK (n=roy@148.122.35.210) |
| 14:20.41 | stoffell_h | tzafrir_laptop: i'm using bristuff -1q and still have the infamous HANGUP issue, anything I can do to send debug log to Kapejod ? |
| 14:21.38 | *** join/#asterisk iax (i=mbrooks@hijacked.us) |
| 14:23.35 | *** join/#asterisk NoRemorse (n=bah@210-84-10-183.dyn.iinet.net.au) |
| 14:23.37 | NoRemorse | hi all |
| 14:24.02 | NoRemorse | has anyone heard of an avaya 8300 talking to asterisk using h.323 ? |
| 14:24.09 | stoffell_h | tzafrir_laptop: cancel my previous request... the bug is gone.. my mistake! |
| 14:25.34 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 14:25.42 | *** join/#asterisk Hmmhesays (n=Hmmhesay@66.173.103.110) |
| 14:25.45 | Hmmhesays | weehaw |
| 14:25.52 | Damin | Anyone got a PSTN routable number that I can call and just listen to hold music for 30 minutes? |
| 14:25.59 | Hmmhesays | lol |
| 14:26.01 | Damin | Doesn't matter what it is.. |
| 14:26.18 | Hmmhesays | paypal me 10 bucks and sure |
| 14:26.21 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.163) |
| 14:26.47 | FlatFoot | call in or out from |
| 14:27.04 | FlatFoot | damin got 1 in the uk |
| 14:27.14 | Inkubot | thanks guys for your help |
| 14:27.14 | anthm | Damin how bout iowa |
| 14:27.15 | Inkubot | bye |
| 14:27.24 | Hmmhesays | I got one in jersey, but I want some cash to use it |
| 14:27.26 | syzygybsd | Damin: just call popcorn |
| 14:27.35 | Damin | anthm: Iowa is good! :) |
| 14:27.44 | anthm | call our dev conference it's an iowa did |
| 14:27.53 | Damin | anthm: Alright.. |
| 14:27.57 | Damin | anthm: DID? |
| 14:27.57 | E-bola | heys guys, i got a normal setup with a sip->pstn provider |
| 14:28.00 | anthm | 712-432-7800 |
| 14:28.05 | E-bola | when ppl dial in i can chose which local extensions ring |
| 14:28.05 | syzygybsd | wait.. popcorn hangs up automatically after 1 minute... |
| 14:28.14 | E-bola | but how can i make external cellphones and stuff ring too? |
| 14:28.51 | syzygybsd | E-bola: dial(sip/exten&&Zap/g1/cellnumber) |
| 14:29.40 | *** join/#asterisk JackEStorm (n=thinkthi@ip68-225-72-125.no.no.cox.net) |
| 14:31.14 | E-bola | i dont have zap |
| 14:31.22 | *** join/#asterisk froguz (n=xxxxx@pc-95-155-104-200.cm.vtr.net) |
| 14:31.23 | E-bola | i call pstn via my sip>pstn provider |
| 14:31.39 | Dovid | where can I get polycom updates for the IP 601 ? |
| 14:31.50 | E-bola | cant i make the dialplan act like if a user picks up his ipphone and dials a pstn number |
| 14:32.26 | *** join/#asterisk anonymouz666 (n=anonymou@20151155235.user.veloxzone.com.br) |
| 14:32.53 | *** join/#asterisk lusum (n=lusum@host-84-221-87-158.cust-adsl.tiscali.it) |
| 14:32.53 | syzygybsd | E-bola: it is the same thing I said up there |
| 14:33.02 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:33.21 | syzygybsd | dial(sip/exten&&sip/cellnumber@provider) |
| 14:33.34 | E-bola | hmm ya i tried that |
| 14:33.34 | *** join/#asterisk ncjp (n=switch@61.206.115.5.user.ad.il24.net) |
| 14:33.40 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 14:33.43 | E-bola | i think its looking for cellnumber in my own domain |
| 14:33.52 | syzygybsd | Dovid: you need to find a polycom reseller |
| 14:33.56 | *** part/#asterisk lusum (n=lusum@host-84-221-87-158.cust-adsl.tiscali.it) |
| 14:34.46 | pdtmobile | Dovid: what you need? |
| 14:35.06 | froguz | i'm trying to dial over a micronet fxo gateway, with sip account 1001, so when i dial that number from asterisk it gives me dial tone on fxo interface, but i can't dial any number (is not responding to dtmf's) |
| 14:35.44 | froguz | exten => 1001,1,Dial(SIP/1001,,tT) <- that's the way i get into de fxo. |
| 14:36.09 | froguz | what could be the problem? |
| 14:36.27 | syzygybsd | froguz: just a guess, but you probably have to send the whole number to start with. |
| 14:36.57 | syzygybsd | Dial(SIP/mynumber@1001,,tT) |
| 14:37.02 | E-bola | syzygybsd: when u said @provider what should iu replace provider with? |
| 14:37.05 | E-bola | the realm ? |
| 14:37.17 | E-bola | or the definition in sip.conf name |
| 14:37.27 | syzygybsd | definition in sip.conf |
| 14:37.37 | froguz | syzygybsd, i'm goint to try that |
| 14:38.12 | E-bola | trying... |
| 14:38.13 | syzygybsd | froguz: it probably isn't sending the DTMF tones because it never got an answer from the other end |
| 14:41.04 | E-bola | great works syzygybsd |
| 14:42.28 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.8.Dial1.SanJose1.Level3.net) |
| 14:43.28 | *** join/#asterisk RoyK (n=roy@148.122.35.210) |
| 14:46.51 | *** join/#asterisk zaptel (n=just@nat1.inalambrica.net) |
| 14:47.26 | *** join/#asterisk jonconley (n=jonconle@smtp.Nemschoff.com) |
| 14:48.48 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 14:48.51 | Dovid | can i use this as a catch all account fo VM ? set one box to _XX ? |
| 14:49.03 | E-bola | do any1 use asterix from mobile devices? |
| 14:49.10 | E-bola | im looking for a good client for windows mobile 5.0 |
| 14:49.37 | Dovid | yes |
| 14:49.40 | Dovid | there are soft phones |
| 14:50.15 | E-bola | know any? |
| 14:50.20 | E-bola | ohh sory i meant a free one |
| 14:50.28 | E-bola | would like to try it without having to buy something |
| 14:50.30 | Dovid | lol |
| 14:50.33 | Dovid | not really |
| 14:50.38 | Dovid | there is sjphone i believe |
| 14:50.47 | Dovid | its $30.00 USD |
| 14:50.50 | Dovid | from xten |
| 14:51.21 | Nugget | xlite or sjphone are both free. sjphone is more solid, but xlite is closer to something you could give to a user. |
| 14:51.29 | syzygybsd | xten has a free one too, not sure how well it works on mobile devices |
| 14:52.03 | E-bola | u sure xlite is free for mobile devices too? |
| 14:54.12 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 14:54.13 | *** join/#asterisk bikola_c (n=bikola@217.056.dsl.brs.iprimus.net.au) |
| 14:54.46 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.8.Dial1.SanJose1.Level3.net) |
| 14:55.06 | *** join/#asterisk hohum (n=dcorbe@mail.interceltelecoms.com) |
| 14:55.21 | syzygybsd | I have never used it for mobile devices |
| 14:55.45 | E-bola | sjphone has one for mobile devices |
| 14:55.50 | E-bola | only a 30 day trial though |
| 14:55.55 | E-bola | but guess thats neough to test it |
| 14:56.52 | froguz | syzygybsd, i've noticed that every time i send a digit to the fxo gateway, asterisk cli show me this: Attempting native bridge of SIP/3000-c43b and SIP/1001-1f7e |
| 14:57.20 | syzygybsd | k... you must be calling out of extension 3000 |
| 14:57.28 | froguz | but dialtone keep on running. |
| 14:57.52 | froguz | yeap, i'm calling from 3000(x-lite) to 1001(FXO gateway) |
| 14:57.52 | syzygybsd | froguz: you didn't try to send the whole dial string at once yet have you? |
| 14:58.05 | froguz | yes i have |
| 14:58.11 | froguz | but nothing happends |
| 14:58.22 | syzygybsd | paste the line from your extensions.conf please |
| 14:58.28 | froguz | ok |
| 14:58.46 | froguz | exten => _1001.,1,Dial(SIP/${EXTEN:4}@1001,,tT) |
| 14:58.46 | froguz | exten => 1001,1,Dial(SIP/1001,,tT) |
| 14:59.04 | froguz | that give me the option to try both methods |
| 14:59.12 | syzygybsd | k, then second one will pick it up. remove it |
| 14:59.29 | syzygybsd | the second one wont' work |
| 14:59.31 | froguz | when i'm working with zap channels, that both ways work |
| 14:59.35 | *** join/#asterisk logicalmark (n=root@adsl-065-015-150-223.sip.bhm.bellsouth.net) |
| 14:59.42 | syzygybsd | this isn't a zap channel |
| 14:59.48 | froguz | aha |
| 15:00.07 | froguz | but the first line doesn't work neither |
| 15:00.32 | logicalmark | Hey guys. What disto would you say is the easiest to use when installing asterisk? |
| 15:00.41 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 15:00.48 | syzygybsd | did you have both of them in the same context at the same time? |
| 15:00.52 | Qwell | logicalmark: Whatever you're confortable with |
| 15:00.58 | jbroome | i install * from source, so it doesn't really matter which distro i use |
| 15:01.03 | RoyK | fuck. i should have had a really good c teacher. |
| 15:01.06 | syzygybsd | logicalmark: I like debian |
| 15:01.07 | jbroome | whichever is easiest to install the compilers. :) |
| 15:01.13 | froguz | syzygybsd, yes |
| 15:01.23 | drray | RoyK, you would ahve quit earlier |
| 15:01.23 | MrChimpy | royk: no substitute for a really good book |
| 15:01.34 | Hmmhesays | I want candy |
| 15:01.35 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 15:01.37 | bikola_c | hey guys, what are the maximum number of simultaneous calls which can be made asterisk@home |
| 15:01.38 | syzygybsd | right.. well remove the second one. They are likely to be causing problems with eachother |
| 15:01.50 | drray | I never used asterisk@home |
| 15:01.51 | jbalcomb | bikola_c: 65,535 |
| 15:01.54 | syzygybsd | now get in the van |
| 15:01.56 | logicalmark | jbroome: Qwell: and syzygybsd: I am new to linux and do not know which is easiest to install the compilers or which I am more comfortable with. |
| 15:01.59 | Hmmhesays | I dumped my bike this weekend |
| 15:02.01 | Hmmhesays | its toast |
| 15:02.02 | RoyK | just trying to extract some sip headers.... |
| 15:02.08 | drray | you ok? |
| 15:02.12 | *** join/#asterisk ivanfm (n=ivanfm@201.52.162.52) |
| 15:02.14 | jbroome | logicalmark: hell, use ubuntu, that should be easy enough |
| 15:02.26 | Hmmhesays | superficial wounds |
| 15:02.31 | Hmmhesays | lots of pain, little damage |
| 15:02.33 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 15:02.44 | jbalcomb | He's on IRC from the hospital. they made special cuts in his body cast to allow for typing. =) |
| 15:02.44 | Hmmhesays | lots of damage to the bike |
| 15:02.47 | jbroome | chicks dig scars |
| 15:02.57 | *** part/#asterisk logicalmark (n=root@adsl-065-015-150-223.sip.bhm.bellsouth.net) |
| 15:03.02 | Hmmhesays | kicks dig guitarist who ride motorcycles and are computer geeks |
| 15:03.05 | Hmmhesays | I'll tell you what |
| 15:03.06 | tzanger | who's in the hospital? |
| 15:03.17 | syzygybsd | Hmmhesays: |
| 15:03.20 | Hmmhesays | no I'm not in the hospital |
| 15:03.25 | bikola_c | jbalcomb: are you sure, after attempting to reseatch a little, i find there is asterisk business edition which boasts about being able to make simultaneous calls, i thoguht doesnt a@home |
| 15:03.28 | Hmmhesays | although I did experience a whole new level of pain this weekend |
| 15:03.39 | syzygybsd | good pain killers? |
| 15:03.47 | froguz | syzygybsd: this is what i get when i dial 10012885125: http://pastebin.ca/90269 |
| 15:03.53 | Hmmhesays | ibuprofen and acetaminophen |
| 15:04.13 | froguz | wich should dial 2885125 over the fxo gateway |
| 15:04.22 | *** join/#asterisk hohum (n=dcorbe@12.195.58.235) |
| 15:04.52 | *** join/#asterisk dwrecktion (n=dwreckti@71.16.158.170) |
| 15:04.58 | *** join/#asterisk momelod (n=momelod@bas5-toronto12-1128748735.dsl.bell.ca) |
| 15:05.04 | momelod | hello people |
| 15:05.15 | Dovid | hello |
| 15:05.20 | dwrecktion | has anybody implemented an overhead paging system with their asterisk box? |
| 15:05.21 | Dovid | anyone getting thsi respons from teliax ? |
| 15:05.22 | Dovid | v |
| 15:05.26 | Dovid | chan_iax2.c:2860 auto_congest: Auto-congesting call due to slow response |
| 15:05.28 | Dovid | yes I have |
| 15:05.37 | syzygybsd | well.. the Tt is in the wrong place... |
| 15:05.50 | dwrecktion | Dovid: you have implemented an overhead paging system? |
| 15:05.53 | syzygybsd | nm.. was putting another , in there |
| 15:05.53 | Dovid | dwrecktion: i have a phone that has auto answer and does it |
| 15:06.08 | Dovid | PM me |
| 15:06.44 | froguz | what's wrong with de Tt?? |
| 15:07.03 | syzygybsd | froguz: i don't know, it seems strange that it is bridging all those times, could there be another problem? |
| 15:08.23 | froguz | i think it is bridging once per each extra number i send to the extension (i.e. the phone number) |
| 15:08.26 | syzygybsd | nothing, I was putting it in the wrong place in my head... |
| 15:08.36 | syzygybsd | though just for testing I would take out all options |
| 15:08.51 | froguz | le me see |
| 15:09.08 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:09.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:09.39 | [TK]D-Fender | Hmmhesays : You're a regular Jimmy Dean.... |
| 15:09.58 | Qwell | mmm...sausage... |
| 15:10.16 | fholmes | Has anyone here setup agent queues with an auto attendant before? I want to know the correct context for my agent extensions in my dial plan. Under the [default] context, their own context or under the queue context? |
| 15:10.58 | syzygybsd | fholmes: I think i have.. |
| 15:11.11 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:11.19 | Hmmhesays | [TK]D-Fender, haha james dean was way cooler than me |
| 15:11.21 | syzygybsd | the agent extension should be in their own context |
| 15:11.54 | syzygybsd | that way they can dial whoever they want.. IE other agents, internal lines, external lines... |
| 15:12.01 | *** join/#asterisk pdtmobile (n=ptinsley@adsl-157-59-139.bna.bellsouth.net) |
| 15:12.09 | syzygybsd | you don't want them to dial into the queue when they pick up the phone do you? |
| 15:12.16 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:12.16 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:12.45 | fholmes | syz: so how should I reference them in my script? Do I need to do an include in the [default] context or just in my auto attendant script do a goto with the extension and the correct context? |
| 15:13.02 | fholmes | No I don't want the to dial into the queue when they dial out. That is for sure. |
| 15:13.21 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 15:13.42 | syzygybsd | k, in your sip.conf just setup the context=myagentcontext |
| 15:15.18 | fholmes | So if I have in my agent script the following: exten => s,n,WaitExten and someone dials say 1201 (one of my agents extensions) it will go to that extension even if it is not in the default context? |
| 15:15.32 | fholmes | agent script = Auto attendant script. |
| 15:15.59 | momelod | quick question, i have 5 fxo lines, how could i setup my extensions.conf so that it tries dialing out on the first available line? |
| 15:16.14 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:16.45 | syzygybsd | well, it depends on how you have it setup, but it doesn't matter where in your extensions.conf you put the dial command it will get sent to the right extension |
| 15:16.45 | momelod | i tried w/ Zap/1&Zap/2&Zap/3... but instead of dialing, i just get a new dialtone |
| 15:17.05 | syzygybsd | momelod: setup a zap group, then dial zap/g1 |
| 15:17.43 | momelod | syzygybsd: that sounds like what i need, ..to group my zap channels |
| 15:18.03 | syzygybsd | it is in your /etc/asterisk/zaptel.conf file |
| 15:18.26 | momelod | ahhh, thanx so much |
| 15:19.38 | syzygybsd | I have beat my head over a fair share of asterisk problems, it is nice to be of help |
| 15:19.55 | froguz | what is the correct dtmf payload setting? |
| 15:19.58 | sevard | everyone has that bloody welt. |
| 15:21.28 | syzygybsd | ya, tis sad when it is because of my clients distro choices though |
| 15:21.37 | Dovid | whats the cheapest IP phone out there that ahs auto answer ? |
| 15:21.50 | *** join/#asterisk tclark (n=TC@S0106000f66c5d294.gv.shawcable.net) |
| 15:22.03 | sevard | syzygybsd: i say "slackware or not at all." |
| 15:22.06 | syzygybsd | asterisk is FREE! |
| 15:22.21 | syzygybsd | sevard: I really like debian stable |
| 15:22.28 | mog | ewwwwwwwwww slackware aka "the best 1995 had to offer" |
| 15:22.35 | sevard | i've never really dipped into debian, but i've heard a lot about it. |
| 15:22.42 | syzygybsd | lol.. ya, about when I stoped with slackware |
| 15:22.48 | [TK]D-Fender | sevard : I typically prefer Slackware myself, but have gotten to liking CentOS a fair bit. FC blows chunks. |
| 15:22.55 | jbroome | indeed |
| 15:22.58 | sevard | I really really really really hate CentOS |
| 15:23.03 | sevard | truly. |
| 15:23.05 | mog | centos = fc = redhat = crap |
| 15:23.07 | syzygybsd | it is nice because the main thing they want is stable. and for any server that is what I want |
| 15:23.13 | mog | debian and derivs. are the way to go |
| 15:23.20 | syzygybsd | I dont' care about the latest and greatest, but stable... |
| 15:23.32 | mog | debian stable = most stable thing ever |
| 15:23.33 | mog | old |
| 15:23.36 | mog | but stable as shit |
| 15:23.42 | syzygybsd | exactly mog |
| 15:23.43 | mitcheloc | ;) |
| 15:23.47 | mog | see perfect example |
| 15:23.47 | sevard | syzygybsd: same here, slackware is the only distro i've used that won't crash on me. i've never used debian but i like my slackware. |
| 15:23.55 | mog | mitcheloc is trying to get trunk working on fc |
| 15:23.56 | sevard | it's familar, safe, stable and light. |
| 15:24.02 | mog | he has all these problems with libraries |
| 15:24.05 | mog | that just dont happen in debian |
| 15:24.09 | mitcheloc | mog, is that why your stuffisn't working on centos? =P |
| 15:24.13 | mog | no |
| 15:24.25 | mog | my stuff isnt working as you have not linked libiksemel with the gnutls |
| 15:24.28 | mog | and it is working |
| 15:24.35 | mog | just not working with servers that require tls |
| 15:24.41 | mog | which are 99% these days |
| 15:24.49 | mog | i think jabber.org might not require it for a connection |
| 15:24.52 | syzygybsd | I love apt. I install it on any other box I am working on FC or Centos |
| 15:24.52 | mog | but i could be wrong |
| 15:24.57 | mog | lol |
| 15:25.35 | sevard | see I don't need apt-get for any other reason than a dev box or a desktop where i install and remove applications all the time |
| 15:25.52 | mitcheloc | ahh so there is an option to compile libiksemel with tls.. |
| 15:26.04 | sevard | i have no problems with my servers i build to find the stuff i need and compile it. slackware is pretty straight forward. no hidden scripts like rh ;( |
| 15:26.13 | syzygybsd | is is nice to be able to just "apt-get install asterisk" and have all the depenancies installed, then you can just download and compile the source if you want the latest |
| 15:26.35 | mog | yes |
| 15:26.37 | syzygybsd | sevard: don't do security updates much? |
| 15:26.44 | mog | thats why the error says no gnutls installed... |
| 15:26.50 | sevard | that sounds nice, but it doesn't bother me to find an application and compile it. |
| 15:26.55 | sevard | syzygybsd: on a regular basis |
| 15:27.20 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 15:27.20 | Hmmhesays | so injecting drivers into a windows xp install iso is easy as pie |
| 15:27.30 | mog | actually syzygybsd, you could just say apt-get build-dep asterisk |
| 15:27.30 | syzygybsd | apt is great for those, does slackware have an easy way or do you have to go find them all manually? |
| 15:27.40 | mog | and it would get everything needed to build and run asterisk |
| 15:27.43 | mog | but not asterisk itself |
| 15:27.53 | mog | so then you could really just build and run asterisk |
| 15:28.18 | sevard | syzygybsd: as far as I know I don't know anything about an 'automatically update my operating system for security updates regardless of source just tell me it's updated so i'm happy" sort of tool for slackware. i do all my updates manualy |
| 15:28.55 | mog | you can still do it all manually apt-get upgrade will list what is to be installed |
| 15:28.59 | mog | and then just do it by hand |
| 15:29.01 | syzygybsd | ya, probably better that way for a server enviornment, you dont' want your server dying because some patch |
| 15:29.24 | sevard | automated tools scare me ;/ |
| 15:29.27 | mog | besides i see no more reason to trust slack tarballs than debian .debs |
| 15:29.35 | mog | you can use them manually if you want |
| 15:29.37 | sevard | who said anything about slack packages? |
| 15:29.48 | pdtmobile | sevard: they become a necessity when you manage 100+ linux boxes |
| 15:29.49 | syzygybsd | ya.. he was saying there wasn't an automatic program... |
| 15:29.51 | mog | ugh you have way to much time on your hands |
| 15:29.52 | *** join/#asterisk SimoAmi (n=SimoAmi@pool-70-107-83-214.ny325.east.verizon.net) |
| 15:30.08 | sevard | pdtmobile: I understand that necessity, i don't administeer that amount of boxen, about 20-30 at the moment. |
| 15:30.11 | mog | not enough |
| 15:30.12 | syzygybsd | lucky mog |
| 15:30.26 | syzygybsd | I only have 14... no wait.. I can't count that high.. |
| 15:30.32 | mitcheloc | mog, what kind of special build command is it? it doesn't look documented... it it something like -- "./configure --prefix=/usr --enable=gnutls" |
| 15:30.38 | pdtmobile | I don't either anymore, but last place I worked we had them all over the country and loads of them so any automation you could do was a life saver |
| 15:30.47 | mog | i think it finds it by default |
| 15:30.54 | mog | but ./configure --help tells you |
| 15:30.56 | mog | for sure |
| 15:31.15 | mitcheloc | okay, and is SVN better then their tarbell? |
| 15:31.48 | syzygybsd | I prefer their tarball for the feeling of stability |
| 15:32.01 | jbalcomb | Is port 4569 something to do with IAX and/or Asterisk? |
| 15:32.05 | rob0 | There are various Slackware package automation tools, like slapt-get and swaret. I don't use them, just tossing this out, sorry, I missed the context of the discussion. |
| 15:32.33 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 15:32.59 | syzygybsd | jbalcomb: I have that UDP port open so I am guessing IAX2 |
| 15:33.38 | jbalcomb | oki, thankas |
| 15:34.09 | mog | what mitcheloc ? |
| 15:35.20 | sevard | rob0: ah yes, i remember using slapt-get, but i believe i used it at an early stage of development and it was highly unstable, don't recall using it after. |
| 15:35.30 | mitcheloc | mog, i meant if iksemel svn was a better choice vs the .tar.gz file, i recompiled it and it looked like it found the tls stuff...but asterisk had the same error... i'm recompiling asterisk incase thats needs to be done as well...i'm just not having good luck eh |
| 15:35.34 | [TK]D-Fender | sevard : Swaret upgraded my X and killed it :) |
| 15:35.43 | mog | they are the same basically |
| 15:35.46 | mog | pretty stable package |
| 15:35.54 | *** join/#asterisk morex (i=morex@host86-133-105-226.range86-133.btcentralplus.com) |
| 15:35.54 | sevard | [TK]D-Fender: awesome. |
| 15:35.56 | morex | Hey there |
| 15:36.19 | morex | Any C programmers out there want to help me fix the big manager-queues-deadlock bug? |
| 15:36.22 | [TK]D-Fender | sevard : So I'm either going to upgrade to 11 (when Pat gets around to it), or likely CentOS. |
| 15:36.32 | FlatFoot | general question are most of you online today from the america's ? |
| 15:36.36 | sevard | the closest I've ever gotten to using Debian was DSL (damn small linux) |
| 15:36.50 | sevard | [TK]D-Fender: that bastard is slacking |
| 15:37.05 | [TK]D-Fender | sevard : Feel the irony... |
| 15:37.08 | sevard | :) |
| 15:37.36 | morex | FlatFoot: UK here |
| 15:37.44 | FlatFoot | UK here |
| 15:38.03 | morex | Cool |
| 15:38.10 | FlatFoot | sunny kent |
| 15:38.12 | syzygybsd | there is a fair ammount from random places |
| 15:38.14 | morex | Sunny Devon |
| 15:38.21 | syzygybsd | lol.. poor kent.. |
| 15:38.23 | FlatFoot | bleedin ot init |
| 15:38.29 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 15:38.32 | morex | LOL |
| 15:38.38 | TripleFFFF | hey wahts the app command to switch to ulaw |
| 15:38.41 | *** join/#asterisk CoffeeIV (i=rgr@cpe-70-112-100-20.austin.res.rr.com) |
| 15:38.42 | TripleFFFF | or any codec.. |
| 15:38.43 | morex | Phew wha a scorcher! |
| 15:38.55 | syzygybsd | is it true what they say about the girls from kent? |
| 15:38.58 | FlatFoot | sposed to get 35 2moz |
| 15:39.05 | FlatFoot | the girls ( BIKES ) |
| 15:39.07 | morex | Right I guess there are no l33t c hackers on this channel |
| 15:39.27 | morex | Anybody know how to do a producer/consumer buffered writer in C? |
| 15:40.01 | syzygybsd | what are petrol prices over in the UK right now? |
| 15:40.08 | morex | Silly. |
| 15:40.14 | FlatFoot | 99ppl ish |
| 15:40.29 | syzygybsd | ouch |
| 15:40.30 | FlatFoot | morex: you on the mailing list ? |
| 15:40.35 | morex | FlatFoot: Yup |
| 15:40.43 | syzygybsd | I am sad it is 2.83 over here |
| 15:40.49 | syzygybsd | that is less then half |
| 15:41.04 | morex | That'd be per gallon though... |
| 15:41.05 | FlatFoot | there is a bloke name of Darren who did some libpri work for me so i could get Type5 auth |
| 15:41.13 | syzygybsd | ya... |
| 15:41.17 | TripleFFFF | anyone ? |
| 15:41.19 | morex | This is per litre |
| 15:41.35 | *** join/#asterisk RoyK (n=roy@148.122.35.210) |
| 15:41.46 | morex | I'll try posting to asterisk-dev I guess |
| 15:41.55 | FlatFoot | morex: Mail in aim it at Darren ( think it was Storer) |
| 15:42.14 | *** join/#asterisk eliel (n=eliel@200.123.183.89) |
| 15:42.22 | FlatFoot | morex: he was if i remember right pretty sh1t hot at c |
| 15:42.40 | TripleFFFF | so theres no way to exten ,1,switchcodec(ulaw) ? |
| 15:42.47 | TripleFFFF | or is it set(codec=ulaw) ? |
| 15:43.05 | syzygybsd | morex: ya, petrol is .40% the cost over here |
| 15:43.13 | syzygybsd | sorry, ignore the . |
| 15:43.20 | morex | FlatFoot: Got an email address? |
| 15:43.30 | FlatFoot | i'll have a look back in a bit |
| 15:43.38 | morex | I don't want to write to the whole list in case I'm wrong |
| 15:43.59 | *** join/#asterisk jero (n=jero@OTWAON23-1177993857.sdsl.bell.ca) |
| 15:44.10 | syzygybsd | ono.. you can't ever be wrong... |
| 15:44.31 | syzygybsd | if you write the list people who have the same question in the future will know |
| 15:45.00 | *** part/#asterisk jero (n=jero@OTWAON23-1177993857.sdsl.bell.ca) |
| 15:45.11 | morex | Syzygybsd: I wouldn't want to pollute it with misinformation for that very reason. |
| 15:45.15 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 15:45.28 | mut | anyone used linux odbc before? |
| 15:45.49 | sevard | So here's a brain teaser. Say a guy has an * box with a PRI and SIP clients are registered under several contexes. How does one give a certian context only two trunks out without grouping them in zapata.conf? I heard mention of a counter in the astriskdb? |
| 15:45.54 | syzygybsd | morex: I look at peoples answeres more then the question |
| 15:46.08 | Hmmhesays | why not use zapata.conf? |
| 15:46.28 | CoffeeIV | Where should you get the app_rxfax.c and app_txfax.c for spandsp from ? I got the latest spandsp, but those files weren't there, and when I got them from an older spandsp, it didn't compile |
| 15:46.40 | sevard | Hmmhesays: because if you limit "2 for this guy" in zapata.conf you can't 'oversell' your trunks |
| 15:47.03 | sevard | they would be dedicated trunks |
| 15:47.12 | morex | syzygybsd: What's needed here is a patch. |
| 15:47.19 | stoffell_h | CoffeeIV: use an older version that HAS the files.. (that's the recommended way!) |
| 15:47.26 | Strom_C | sevard: that sounds like really bad traffic engineering to me |
| 15:47.34 | sevard | how os |
| 15:47.34 | Hmmhesays | use gotoifs |
| 15:47.36 | sevard | +so |
| 15:47.48 | syzygybsd | sevard: you could do a DB(guyscalls=$[${guyscalls}+1]) then the same afterwards |
| 15:48.04 | Strom_C | sevard: go play with the erlang formulas |
| 15:48.21 | syzygybsd | but that could cause problems if asterisk dies incorrectly |
| 15:48.23 | CoffeeIV | stoffell_h: ok, I can do that -- but what's the point of the newer versions of spandsp then ? non-asterisk fax applications ? |
| 15:48.59 | syzygybsd | sevard: wait.. dedicated trunks? |
| 15:49.18 | sevard | I don't know whath erlang formulas and I see the problem if asterisk died, but effectivly you could zero out a counter on boot |
| 15:49.29 | sevard | syzygybsd: they'd have to be 'dynamic' trunks, non dedicated |
| 15:49.35 | Hmmhesays | bahwittaba |
| 15:49.42 | Strom_C | sevard: erlang formuas are crucual to traffic engineering |
| 15:50.05 | TripleFFFF | CoffeeIV if you find out let me know |
| 15:50.12 | TripleFFFF | CoffeeIV got same problem |
| 15:50.36 | Strom_C | god |
| 15:50.49 | Strom_C | my spelling is atrocious immediately after waking up |
| 15:50.50 | sevard | yes, my son? |
| 15:50.55 | syzygybsd | sevard: so do you want to limit incomming or outgoing? |
| 15:51.07 | Strom_C | erlang formulas are crucial to traffic engineering |
| 15:51.27 | sevard | syzygybsd: well, both, but I'm starting with outgoing |
| 15:51.33 | syzygybsd | i like crucual better |
| 15:51.37 | Strom_C | I would strongly recommend that if you're doing anything on even a medium scale, you look at that formula |
| 15:51.38 | sevard | it might just end up to be outgoing. |
| 15:51.48 | *** join/#asterisk esculapio__ (n=ESCulapi@200.88.44.66) |
| 15:52.43 | TripleFFFF | cofee ? |
| 15:53.09 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 15:53.20 | syzygybsd | sevard: just a thought.. in your sip.conf "call-limit = number : Number of simultaneous calls through this user/peer." |
| 15:53.21 | sevard | Strom_C: I understand that model, I can adjust to adverages once i know how to limit outgoing trunks |
| 15:53.43 | sevard | syzygybsd: but how about a group of users? |
| 15:53.49 | Strom_C | sevard: you don't limit it at the trunk level, you limit it at the user level |
| 15:53.52 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 15:54.01 | sevard | + i only want to limit calls on the PRI not on SIP or IAX |
| 15:54.09 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 15:54.15 | Strom_C | sevard: what? |
| 15:54.17 | Strom_C | sevard: why? |
| 15:54.34 | sevard | because SIP and IAX calls are inhouse and 'free' |
| 15:54.54 | CoffeeIV | TripleFFFF: I'll pm you or post here when I find a version of spandsp/rxfax/txfax that works |
| 15:55.03 | TripleFFFF | yeah |
| 15:55.06 | TripleFFFF | last one i got compiled |
| 15:55.10 | TripleFFFF | but cores on rxfax |
| 15:55.16 | syzygybsd | if they are inhouse make them getup from the desk and go talk to the person... lol |
| 15:55.21 | CoffeeIV | which asterisk are you using ? |
| 15:55.24 | sevard | big house :) |
| 15:55.26 | TripleFFFF | 2.9.1 |
| 15:55.33 | anonymouz666 | Does Linksys PAP2 support 2 calls using G729? |
| 15:55.41 | Strom_C | sevard: please, dude, don't make me beat you over the head with a phone |
| 15:55.42 | TripleFFFF | pap2-t they say |
| 15:55.44 | TripleFFFF | anonymouz666 |
| 15:56.03 | sevard | but anyway, even if i have to lump zap/sip/iax all together i dont' want to limit calls on an individual basis, i want to limit the amount of zap calls going out per group of users. |
| 15:56.27 | Strom_C | sevard: what you're doing is supremely bad traffic engineering |
| 15:56.46 | syzygybsd | Strom_C: I can see the business need for it though |
| 15:56.48 | *** join/#asterisk znoG (n=gs@162-148-235-201.fibertel.com.ar) |
| 15:57.05 | sevard | syzygybsd: strom likes to tell people how to run their lives ;/ |
| 15:57.08 | syzygybsd | It could be horrible to setup that way, but if that is how he bills for it... |
| 15:57.16 | sevard | i don't bill anybody. |
| 15:57.19 | *** join/#asterisk dtwilson (n=dtwilson@alfa0002.autosols.com) |
| 15:57.33 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 15:57.33 | syzygybsd | oh, then why the limit? |
| 15:57.35 | CoffeeIV | TripleFFFF: same here (if you mean 1.2.9.1 that is) -- I am now trying the spandsp-0.0.2pre26 |
| 15:57.37 | sevard | everything is the man upstairs, slowly paing for my school :) |
| 15:57.39 | Strom_C | sevard: no, I'm warning you that you're wasting your time on something that will only cause you headaches in the future |
| 15:57.46 | TripleFFFF | yeah |
| 15:57.58 | TripleFFFF | spandsp-0.0.2pre26.tar.gz |
| 15:58.02 | TripleFFFF | spandsp-0.0.3pre22.tgz |
| 15:58.06 | TripleFFFF | bot tried |
| 15:58.08 | *** part/#asterisk dtwilson (n=dtwilson@alfa0002.autosols.com) |
| 15:58.26 | sevard | so, syzygybsd, an ideas? |
| 15:58.28 | sevard | any* |
| 15:58.28 | syzygybsd | sevard: does the man upstairs listen? |
| 15:58.30 | *** join/#asterisk salviadud (n=ralfalfa@dsl-201-128-132-150.prod-infinitum.com.mx) |
| 15:58.35 | TripleFFFF | ls -la app_rxfax.c |
| 15:58.35 | TripleFFFF | -rw-r--r-- 1 root root 13190 Feb 7 19:58 app_rxfax.c |
| 15:58.38 | syzygybsd | oh.. I could do it in 15 minutes... |
| 15:58.39 | sevard | syzygybsd: not to a single word and I'm past caring. |
| 15:58.45 | TripleFFFF | also what os you onto CoffeeIV |
| 15:58.55 | sevard | the mother can go bankrupt |
| 15:59.16 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:59.16 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:59.27 | CoffeeIV | TripleFFFF: CentOS release 4.3 |
| 15:59.29 | sevard | catch me? :) |
| 16:00.15 | *** join/#asterisk N3WWN (n=N3WWN@ns1.futuretek.cx) |
| 16:00.30 | TripleFFFF | CoffeeIV me too |
| 16:00.31 | TripleFFFF | hmm |
| 16:01.14 | N3WWN | I'm having trouble getting the "s" extension to work in the default context. Is there some magic to it? :) |
| 16:01.20 | Hmmhesays | mozilla.com/firefox give me a 403 forbidden |
| 16:01.20 | Hmmhesays | wow |
| 16:02.36 | syzygybsd | gimme a sec, writing |
| 16:04.24 | SimoAmi | verizon is so disappointing |
| 16:04.37 | syzygybsd | the syntax could be a bit off, but http://pastebin.ca/90333 should work |
| 16:04.55 | *** join/#asterisk Setti (n=dfgdfg@145-72-112.adsl.terra.cl) |
| 16:05.03 | CoffeeIV | TripleFFFF: I got the same error compiling app_rxfax.c as before, but you indicated that it compiled for you, but core dumped |
| 16:05.10 | syzygybsd | infact priority 3 needs a ] after 2 |
| 16:05.36 | CoffeeIV | TripleFFFF: I'm going to pastebin the error I am getting |
| 16:05.43 | syzygybsd | but it is very inportant you keep my variable name.. |
| 16:05.45 | syzygybsd | lol |
| 16:06.06 | N3WWN | I have "exten => s,1,Goto(Winbeam,240,1)" in the default context, but get "Extension '7246891239' in context 'default' from '7242190400' does not exist. Rejecting call on channel 0/1, span 1" when I try to call in |
| 16:06.31 | syzygybsd | N3WWN: ya, that is because the number being transfered in there already has an extension |
| 16:06.34 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 16:06.42 | syzygybsd | the s extension is for calls with no extension |
| 16:07.16 | [TK]D-Fender | N3WWN : BRI/PRI does not use "s". All incoming calls HAVE a specified target |
| 16:07.47 | N3WWN | ah... any way to emulate "s"? Or do I need to specify each DID? |
| 16:07.55 | syzygybsd | throw a exten => _X.,1,Goto(Winbeam,240,1) |
| 16:08.02 | sevard | syzygybsd: I understand that it may be a bad idea but I still need to go through with it. Can you point me in the right direction? voip-info isn't helping much these days. |
| 16:08.13 | syzygybsd | sevard: look at my pastebin |
| 16:08.21 | syzygybsd | about 15 lines ^ |
| 16:08.51 | sevard | syzygybsd: sorry, I thought you were helpign somebody else. |
| 16:08.53 | [TK]D-Fender | N3WWN : use a catchall like _X. |
| 16:09.53 | syzygybsd | [TK]D-Fender: you could write the whole line or at least point to the one i wrote ^ |
| 16:09.55 | N3WWN | Do I need _XXXXXXXXXX to catch the 10 digits? |
| 16:10.05 | syzygybsd | the . catches all digits |
| 16:10.22 | N3WWN | Ah! I see... thought that was a period at the end of the sentence :) |
| 16:10.35 | salviadud | jesus man, the period is... |
| 16:10.41 | salviadud | any number |
| 16:10.46 | sevard | syzygybsd: hey, thanks man :) i'm off to read some docs to find out how the hell / what the hell it's doing |
| 16:10.47 | N3WWN | If I have specific extensions defined, the catch-all will not be matched? |
| 16:10.51 | salviadud | or numbers |
| 16:11.14 | salviadud | right |
| 16:11.23 | syzygybsd | sevard: ya, on line 3 it needs a ] after the 2 |
| 16:11.44 | N3WWN | Great! Thanks TKD, syz & salv! |
| 16:12.07 | sevard | syzygybsd: this is a bit confusing :) |
| 16:12.09 | CoffeeIV | Here is the compile error I am getting with app_rxfax.c and spandsp -- anyone have any suggestions ? http://pastebin.ca/90336 |
| 16:12.11 | syzygybsd | salviadud: doesn't have to be numbers... |
| 16:12.22 | syzygybsd | sevard: na, it makes perfect sense... |
| 16:12.30 | salviadud | well, what else can ya dial with a phone? |
| 16:12.37 | salviadud | letters? |
| 16:12.43 | sevard | syzygybsd: do you have time to explain it? or shall i rtfmnub |
| 16:12.54 | syzygybsd | the h, i,t, and s extensions... |
| 16:13.14 | syzygybsd | as well as # and * |
| 16:13.16 | salviadud | some of those only work when Background is on |
| 16:13.26 | TripleFFFF | CoffeeIV configure without zaptel |
| 16:13.39 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B79EF9.pool.t-online.hu) |
| 16:14.12 | TripleFFFF | or make even |
| 16:14.22 | syzygybsd | sevard: oh.. one more thing is you want the extension to be _X. |
| 16:14.25 | CoffeeIV | TripleFFFF: configure what without zaptel ? The spandsp library ? |
| 16:14.33 | TripleFFFF | the asterisk |
| 16:14.48 | TripleFFFF | look in make file comment the zaptel portion and retry |
| 16:14.51 | TripleFFFF | just to see |
| 16:14.54 | TripleFFFF | if it worls |
| 16:15.02 | CoffeeIV | TripleFFFF: looking into it |
| 16:15.35 | *** join/#asterisk sandra78 (n=aerae@200.106.8.56) |
| 16:19.19 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 16:20.33 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 16:20.42 | CoffeeIV | TripleFFFF: It looks like spandsp 0.0.2pre26 will compile now -- I removed the --prefix=/usr/ I was using int he configure for the spandsp library |
| 16:21.09 | TripleFFFF | oh |
| 16:21.12 | TripleFFFF | hehe |
| 16:21.13 | TripleFFFF | yeah |
| 16:21.20 | TripleFFFF | it wirll run and core dump |
| 16:21.24 | CoffeeIV | TripleFFFF: if you having a crash when you get to rxfax, consider checking if /usr/local/lib is where the spandsp lib is, and if that library is in the ld.conf |
| 16:21.25 | TripleFFFF | coz it wont find lib |
| 16:21.29 | TripleFFFF | even fii tieven startus |
| 16:21.31 | CoffeeIV | right |
| 16:21.32 | TripleFFFF | starts |
| 16:22.12 | *** join/#asterisk RaYmAn-Bx (i=rayman@kbhn-vbrg-sr0-vl201-020.perspektivbredband.net) |
| 16:23.19 | stoffell_h | CoffeeIV: the point of the 'newer' version is for developers,you should use rxfax from svn then... |
| 16:24.44 | *** join/#asterisk nettux (n=NetQUACK@adsl-195.137.136.17.tzm.net) |
| 16:25.00 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 16:25.12 | CoffeeIV | stoffel_h: excuses, excuses. That isn't documented anywhere and you didn't mention it when I first described my problem here. If this works I'll add a note the voip-info wiki. |
| 16:25.43 | MRH2 | anyone user a zaptel card on a dell 2650? |
| 16:26.02 | nettux | Hello. I have the following problem: I am trying to configure 9 accounts from a same sip provider on my * box but since they don't pass the fromuser param all the call are considered to come from the same number/account. How can I avoid that? |
| 16:26.32 | Hmmhesays | route in your dp? |
| 16:27.08 | nettux | Yea but they all end up in the same context being considered all to come from the same calling line! |
| 16:27.11 | xachen | :O FreeSWITCH just got slashdotted! |
| 16:27.17 | Hmmhesays | cool |
| 16:27.30 | stoffell_h | CoffeeIV: no, you're right, it's not always obvious |
| 16:27.45 | MRH2 | I am getting that 'orange LED pci error' with the latest zaptel 1.2 branch |
| 16:28.18 | Hmmhesays | opera 9 in windows is freaking sweet |
| 16:28.34 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.134) |
| 16:28.49 | CoffeeIV | stoffell_h: when I get it working and put it in voip-info, then we'll be able to shout "RTF WIKI" at all the newbs with the same problem ;) |
| 16:29.24 | syzygybsd | nettux: don't your sip providers pass the number that was called? |
| 16:29.42 | nettux | syzygybsd looks like they don't... |
| 16:30.01 | nettux | syzygybsd where do you think I should look for it? |
| 16:30.04 | syzygybsd | nettux: my guess is they do... why do you think they don't |
| 16:30.11 | syzygybsd | ${EXTEN} |
| 16:30.20 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 16:30.20 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 16:30.40 | nettux | ${EXTEN} is always on the same account number although they come from different ones |
| 16:30.44 | *** join/#asterisk SexyKen (n=Ken@c-71-202-149-39.hsd1.ca.comcast.net) |
| 16:30.50 | SexyKen | Hey guys, anyone here have a Plasma? |
| 16:31.04 | SexyKen | Plasma TV |
| 16:31.12 | syzygybsd | what do you mean on the same account number? |
| 16:31.17 | Strom_C | what the hell does that have to do with asterisk, SexyKen? :) |
| 16:31.29 | SexyKen | Nothing :-) You guys are just all around useful :-) |
| 16:31.36 | stoffell_h | CoffeeIV: lol |
| 16:31.56 | stoffell_h | CoffeeIV: i prefer iaxmodem with hylafax for faxing.. just a t ip :d |
| 16:32.23 | CoffeeIV | stoffell_h: I may have to try that out if this doesn't work well |
| 16:32.34 | nettux | syzygybsd I mean that I have 9 numbers, say 300 301 302 303...309 |
| 16:32.36 | syzygybsd | hmm.. should I be disapointed my spam only droped by 90% with this new filter? |
| 16:32.38 | nettux | whichever gets the call |
| 16:32.43 | SexyKen | I have a 1080p plasma hooked up using RGB connectors or whatever they're called....but my TV is only showing in a square instead of fullscreen |
| 16:32.43 | nettux | the EXTEN is 308 |
| 16:33.02 | nettux | Got it? |
| 16:33.06 | syzygybsd | nettux: that is a problem with your SIP provider then |
| 16:33.41 | TripleFFFF | Cofee any luck ? |
| 16:33.49 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 16:33.59 | SpaceBass | howdy |
| 16:34.02 | *** join/#asterisk Kryojenik (n=DOH@24-107-84-247.dhcp.stls.mo.charter.com) |
| 16:34.22 | syzygybsd | SexyKen: my guss is there is a zoom button on your remote, press it |
| 16:34.34 | SexyKen | Well that's not very good. |
| 16:34.37 | SexyKen | Distortion when I do that. |
| 16:34.45 | syzygybsd | lol.. well ya |
| 16:34.48 | [TK]D-Fender | SexyKen : Pick the proper video mode. |
| 16:34.48 | Strom_C | SexyKen: sounds like you're watching an analog Tv signal |
| 16:34.48 | GerbilWrk | If you are calling from one sip phone, to another through an asterisk box, does the call connect and all traffic go from phone to phone, or phone to server to phone? |
| 16:35.08 | SexyKen | TK: What do you mean? |
| 16:35.14 | Strom_C | GerbilWrk: depends on if reinvites are enabled |
| 16:35.17 | syzygybsd | you may not have learned this in school, but square doesn't always equal a rectangle |
| 16:35.23 | TripleFFFF | SexyKen you are in 4:3 switch to 16:9 |
| 16:35.32 | SexyKen | The guide shows up in full screen, but the rest shows square. |
| 16:35.34 | syzygybsd | wait.. yes it does.. I mean a rectangle doesn't always equal a square |
| 16:36.31 | syzygybsd | GerbilWrk: phone server phone |
| 16:36.44 | syzygybsd | or what Strom_C said |
| 16:37.41 | [TK]D-Fender | SexyKen : You need to set your VIDEOCARD to 1080i |
| 16:37.53 | SexyKen | Video card? |
| 16:38.09 | GerbilWrk | Strom_C, so say, reinvites are enabled, which way would it go? |
| 16:38.15 | [TK]D-Fender | SexyKen : You are referring to hooking it up to a computer right? |
| 16:39.05 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 16:39.07 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 16:39.46 | syzygybsd | wait.. a 1080p plasma has RGB connectors? |
| 16:40.23 | [TK]D-Fender | syzygybsd : I'm betting its component video..... |
| 16:40.33 | CoffeeIV | TripleFFFF: after all that, my asterisk just crashed in rxfax |
| 16:40.33 | syzygybsd | ya, that would be my guess too... |
| 16:40.57 | SexyKen | Nope |
| 16:41.10 | Assid | err.. RGB and component are the same |
| 16:41.12 | Assid | arent they |
| 16:41.15 | SexyKen | Nope. |
| 16:41.17 | SexyKen | Well |
| 16:41.17 | SexyKen | Wait |
| 16:41.18 | SexyKen | Yea |
| 16:41.21 | SexyKen | Component |
| 16:41.22 | SexyKen | Yes. |
| 16:41.22 | syzygybsd | SexyKen: so there is one plug for each color? |
| 16:41.26 | SexyKen | Yup |
| 16:41.28 | Assid | cause thats what i use |
| 16:41.31 | Assid | for my xbox |
| 16:41.51 | syzygybsd | I haven't seen that for 10 years |
| 16:42.02 | SexyKen | My TV accepts RGB (regular Monitor connector), DVI, Component, Composite & S-Video |
| 16:42.22 | sevard | syzygybsd: is that db() command supposed to be dbput? |
| 16:42.32 | Assid | err.. monitor connects are called something else tho.. |
| 16:42.35 | Assid | if im not mistaken |
| 16:42.40 | [TK]D-Fender | SexyKen : Well your incoming signal is not 1080i and thats the problem then. |
| 16:42.41 | syzygybsd | sevard: depends on the version of asterisk you are using |
| 16:42.47 | syzygybsd | anything new, no |
| 16:42.47 | SexyKen | CGA rather |
| 16:42.48 | [TK]D-Fender | sevard : both |
| 16:42.49 | SexyKen | I mean |
| 16:42.49 | SexyKen | VGA |
| 16:42.50 | SexyKen | Yes |
| 16:42.51 | SexyKen | :-) |
| 16:42.52 | Assid | yeah |
| 16:42.59 | sevard | syzygybsd: ? |
| 16:42.59 | Assid | i keep forgetting the names |
| 16:43.02 | syzygybsd | sexyken: write |
| 16:43.04 | syzygybsd | on |
| 16:43.05 | syzygybsd | one |
| 16:43.07 | syzygybsd | line |
| 16:43.09 | syzygybsd | please |
| 16:43.11 | Assid | whats the problem you getting anyways? |
| 16:43.31 | SexyKen | Yea seriously, that must be annoying. One line please. |
| 16:43.32 | SexyKen | ;-) |
| 16:43.52 | syzygybsd | sevard: they updated the DB command to use the one I do, if you are using a pre 1.2 version you may have to use dbput |
| 16:44.13 | sevard | Asterisk 1.2.7.1 built by root @ thefan on a i686 running Linux |
| 16:44.18 | Assid | 1080i requires an input of that type.. |
| 16:44.22 | syzygybsd | k, then my command should work |
| 16:44.25 | sevard | it fails |
| 16:44.40 | sevard | show applications describing db doesn't show 'db' |
| 16:44.42 | Assid | actually.. if you are using any device, like a DVD, it sometimes depends on the player too... if its software emulated |
| 16:44.44 | syzygybsd | ya, my syntax probably isn't perfect |
| 16:45.08 | [TK]D-Fender | sevard : Set(myvar=${DB(family/key)}) |
| 16:45.14 | Strom_C | that's because DB is a function |
| 16:45.18 | *** join/#asterisk firejon (n=jellis@206-169-49-105.static.twtelecom.net) |
| 16:45.20 | Strom_C | not an application |
| 16:45.22 | syzygybsd | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+db |
| 16:45.24 | [TK]D-Fender | sevard : Set(DB(family/key)=${myvar}) |
| 16:45.44 | syzygybsd | ya, my syntax wasn't the best, I pulled it outta my ass and didn't check it at all |
| 16:45.47 | Corydon-w | 'show function DB' |
| 16:45.58 | TripleFFFF | CoffeeIV any luck ? |
| 16:46.23 | CoffeeIV | TripleFFFF: not yet -- how do you turn on the full log that should be in /var/lib/asterisk/full ? |
| 16:46.41 | Corydon-w | logger.conf |
| 16:46.50 | froguz | anyone who has configured a micronet FXO gateway to make PSTN calls from asterisk? |
| 16:47.17 | TripleFFFF | ./etc/asterisk/logger.conf |
| 16:47.24 | TripleFFFF | then reload logger on CLI> |
| 16:47.40 | CoffeeIV | thanks, trying agian |
| 16:47.57 | Hmmhesays | Wow having opera read things off the urban dictionary is amusing to my 15 year old humor |
| 16:48.48 | vader-- | do you guys allow like *67 calling out of your offices? |
| 16:48.58 | vader-- | im wondering if i should allow it through my dial plan |
| 16:49.04 | TripleFFFF | CoffeeIVwhats your configure for spandsp .. config line |
| 16:49.08 | syzygybsd | sevard: http://pastebin.ca/90367 |
| 16:49.30 | syzygybsd | i tried to match the syntax more on that one |
| 16:49.37 | TripleFFFF | dont forget |
| 16:49.40 | TripleFFFF | all the expectioon |
| 16:49.48 | TripleFFFF | or youll get ++ on every one of them |
| 16:50.00 | TripleFFFF | like busy, no channel, timeout, |
| 16:50.01 | TripleFFFF | etc |
| 16:51.07 | CoffeeIV | TripleFFFF: ./configure, that's it |
| 16:51.46 | CoffeeIV | no info in the /var/log/asterisk/full log, the last thing it says is that it is trying to run rxfax |
| 16:52.32 | TripleFFFF | hehe and a dump ? |
| 16:52.35 | TripleFFFF | hmm |
| 16:52.43 | TripleFFFF | illl ciompile on laptop here centos4.3 |
| 16:52.48 | TripleFFFF | and gdb it to see |
| 16:54.48 | *** join/#asterisk gcarrillog (n=gcarrill@dsl-201-128-97-89.prod-infinitum.com.mx) |
| 16:55.02 | CoffeeIV | I'm not getting a core file from this, my environment isn't set up right to do it |
| 16:55.14 | TripleFFFF | k |
| 16:55.32 | Assid | err.. i cant brelieve it |
| 16:55.38 | Assid | i forgot how to stop asterisk |
| 16:55.51 | Assid | okay got it |
| 16:56.04 | Nivex | it runs so well, why would you need to stop it? ;) |
| 16:56.37 | sevard | ha |
| 16:56.45 | syzygybsd | there are so many ways to kill asterisk |
| 16:56.55 | syzygybsd | wait.. you just want to stop it... |
| 16:57.53 | Assid | had to change the whole routing and everythuing |
| 16:58.02 | Assid | the bandwith it was on was slow.. had to change to the other router |
| 16:58.28 | Assid | thats what happens when you dont shut it .. ever |
| 16:59.45 | *** join/#asterisk heffernan (n=hwt@195.139.204.157) |
| 17:00.10 | heffernan | hi. i have a bunch of users in sip.conf that i want callers to be able to reach from the outside (through a sip trunk) |
| 17:00.28 | docelmo | and? |
| 17:00.34 | TripleFFFF | compiling |
| 17:00.37 | heffernan | however, these numbers are non-sequential, and i don't want to create a rule for every entry. |
| 17:00.41 | syzygybsd | lol.. I love how people think statements are questions... |
| 17:01.04 | heffernan | and matching them with exten => X.,1,Dial(SIP/${ÉXTEN}) does not work. |
| 17:01.12 | syzygybsd | which numbers? and do the numbers have a 1:1 mapping for the sip extensions |
| 17:01.13 | heffernan | in the [incoming] context. |
| 17:01.24 | docelmo | sigh.. This guy is making things too complicated.. |
| 17:01.26 | heffernan | syzygybsd: yes they do. it's the same number on the inside as on the outside. |
| 17:01.27 | TripleFFFF | paste in pastebin |
| 17:01.41 | syzygybsd | heffernan: you need a _ before x.. ie (_X.,1,Dial()) |
| 17:01.52 | heffernan | syzygybsd: sorry, yeah, i already have that. |
| 17:02.11 | syzygybsd | k. pastebin your sip and extensions.conf |
| 17:02.33 | heffernan | syzygybsd: sure, a sec. |
| 17:03.34 | heffernan | syzygybsd: http://pastebin.ca/90384 |
| 17:03.49 | heffernan | syzygybsd: my sip.conf-users are in realtime, but that shouldn't matter should it. |
| 17:04.34 | syzygybsd | umm.. yes? |
| 17:05.04 | syzygybsd | the fact that I can't see if any of your sip users are [${EXTEN}] is what the problem would be |
| 17:05.56 | heffernan | syzygybsd: what do you mean? |
| 17:06.24 | docelmo | No. Realtime users are only present in the system when they register to you |
| 17:06.33 | docelmo | So if they are not registered you cant send calls to them |
| 17:06.34 | syzygybsd | you are trying to dial sip/${EXTEN} do you have any entries in your sip.conf for [whatever extension you are calling] |
| 17:06.43 | jbalcomb | CunningPike: For the Polycom IP501 I'm setting 'reg.1.displayName', 'reg.1.address', 'reg.1.auth.userId', and 'reg.1.server.1.address'. It's not working, what else do I need? |
| 17:07.19 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:07.20 | heffernan | syzygybsd: i have entries for several extensions 223680011 NULL NULL NULL NULL yes internal ... |
| 17:07.38 | harryvv | sixtel is merging with another company. |
| 17:08.30 | docelmo | sixtel bah |
| 17:08.44 | jbalcomb | sixlet yum |
| 17:08.49 | heffernan | syzygybsd: and my phones can't call each other. it will always use the outbound trunk. |
| 17:09.14 | CoffeeIV | TripleFFFF: how's it going ? |
| 17:09.21 | heffernan | syzygybsd: sorry, i forgot. it's possible to call *in* from the outside, but not internally (between extensions in the same context.) |
| 17:09.32 | heffernan | syzygybsd: that's why i have the exten => 67078840,1,Macro(internal,SIP/${EXTEN}) |
| 17:09.47 | heffernan | syzygybsd: i want to replace one for each line with a single line. |
| 17:10.01 | heffernan | syzygybsd: so that it will always try a local extension before using the outbound trunk. |
| 17:10.13 | jbalcomb | How did I get subscribed to the freeswitch mailing list? |
| 17:10.50 | docelmo | sixtel bough iax.cc |
| 17:10.52 | docelmo | ACK! |
| 17:12.10 | jbroome | SYN! |
| 17:12.20 | heffernan | syzygybsd: and exten => _X.,1,macro(internal,SIP/ ... does not work. |
| 17:12.36 | heffernan | FIN! |
| 17:13.40 | syzygybsd | heffernan: k, I have no clue what extensions you have setup but it doesn't look like you have the right ones setup. |
| 17:13.50 | [TK]D-Fender | jbalcomb : Don't need AuthID, just reg.X.address |
| 17:13.52 | *** join/#asterisk [andromeda] (i=formetop@c-24-98-51-95.hsd1.ga.comcast.net) |
| 17:13.57 | harryvv | doc, interesting I thought thay were the same company |
| 17:14.09 | heffernan | syzygybsd: i want a dialplan without any (or at least few) numbers. |
| 17:14.30 | syzygybsd | do you have an extension [67078840] |
| 17:14.38 | heffernan | syzygybsd: i just want the extensions in my database to be able to call each other. |
| 17:14.45 | heffernan | syzygybsd: yes. in the database. |
| 17:15.11 | heffernan | but when i try to call from e.g. 67078840 to one of the other extensions, it uses the outbound trunk. |
| 17:15.18 | jbalcomb | [TK]D-Fender would having AuthID stop it from working? |
| 17:15.20 | [andromeda] | Hi, I am trying to install the asterisk-addons package, so that I will be able to use the MySQL command in my dialplan. However, I get the following errors when I try to run make: |
| 17:15.21 | [andromeda] | cdr_addon_mysql.c:38:19: error: mysql.h: No such file or directory |
| 17:15.40 | [TK]D-Fender | jbalcomb : no, its jsut not necessary. there is a flag to even USE authID. |
| 17:15.44 | heffernan | syzygybsd: that is, if i remove the exten => 67078840,1,M...-lines. |
| 17:15.47 | [andromeda] | I looked in the package I downloaded from the site, and mysql.h is nowhere to be found. |
| 17:15.49 | heffernan | syzygybsd: (which is what i want.) |
| 17:15.53 | [TK]D-Fender | jbalcomb : which is by default = NO. |
| 17:16.14 | syzygybsd | heffernan: just a guess, but you have "include => in" shoudlnt' that be inbound? |
| 17:16.35 | jbalcomb | [TK]D-Fender ok. when it d/l the config it shows the line one button as 4444 but when i press buttons on the phone to see the sip server, it show no entry. what gives? |
| 17:17.23 | heffernan | syzygybsd: no, that is the [in] context. with CFWD, etc. |
| 17:17.26 | [TK]D-Fender | jbalcomb : delete the <mac>-phone.cfg file to make sure the phone isn't overriding the server |
| 17:17.27 | syzygybsd | that and you inbound and outbound contexts have the same extension. So they will overwrite eachother |
| 17:17.40 | syzygybsd | k, i misses that |
| 17:17.46 | [TK]D-Fender | jbalcomb : pastebin a asample |
| 17:17.55 | heffernan | syzygybsd: will they? how? |
| 17:18.31 | syzygybsd | nm, I was thinking you wanted both inbound and outbound, ignore that |
| 17:18.32 | jbalcomb | [TK]D-Fender there is no <mac>-phone.cfg cause my ftp uploads don't work. :/ |
| 17:19.05 | [TK]D-Fender | jbalcomb : :/ |
| 17:19.21 | syzygybsd | heffernan: there is no way to do what you want that i can think of. The random numbers of your phones makes having an entry for each one needed |
| 17:19.22 | [TK]D-Fender | jbalcomb : that makes debuggin kinda hard... |
| 17:19.26 | harryvv | crap iran is selling missles to lebanon |
| 17:19.53 | syzygybsd | lol.. come on, I saw wwIII starting 2 weeks ago |
| 17:19.54 | heffernan | syzygybsd: what about Local/ or something? |
| 17:20.02 | jbalcomb | [TK]D-Fender http://pastebin.ca/90399 |
| 17:20.45 | heffernan | syzygybsd: are you sure about this? that does not make much sense to me. |
| 17:20.50 | syzygybsd | heffernan: k, the phone numbers for your extensions are random. Hence asterisk has no way of knowing which to send local and which to send external unless each one is listed |
| 17:21.19 | GerbilWrk | heffernan, I assume in your sip.conf, the phones have context=internal ? |
| 17:21.24 | [TK]D-Fender | jbalcomb : reg.1.lineKeys="" reg.1.callsPerLineKey="" <-bad |
| 17:21.28 | heffernan | syzygybsd: well, it *could* just check if the number is reachable locally, in the same context, if not, move on and try through the outbound trunk. |
| 17:21.36 | heffernan | syzygybsd: at least that makes sense to me. |
| 17:21.40 | [TK]D-Fender | jbalcomb : and reg.1.lineKeys="" reg.1.callsPerLineKey="" <- 5060 |
| 17:21.51 | syzygybsd | k. there would be a delay though... |
| 17:22.08 | heffernan | GerbilWrk: actually, they have nittedalsnettet, but [nittedalsnettet] includes => internal |
| 17:22.14 | syzygybsd | exten => _X.,1,Dial(sip/${EXTEN}) |
| 17:22.23 | heffernan | GerbilWrk: does that make any difference? |
| 17:22.49 | syzygybsd | exten => _X.,101,Dial(${OUTBOUNDTRUNK}/${EXTEN}) |
| 17:23.42 | GerbilWrk | shouldn't |
| 17:24.38 | heffernan | what is the Local/ good for then? |
| 17:25.39 | *** join/#asterisk glogic (n=gml@66.230.159.3) |
| 17:25.40 | glogic | hi |
| 17:25.48 | *** join/#asterisk veepster (n=veepster@67.130.38.2) |
| 17:25.51 | glogic | i could use some help with my zaptel card |
| 17:26.02 | glogic | is there a way to take a specific channel in and out of service? |
| 17:26.05 | glogic | is there a way to take a specific channel in and out of service?? |
| 17:26.09 | glogic | like say the d-channel? |
| 17:26.27 | syzygybsd | something completly differnet then you want it for |
| 17:26.42 | syzygybsd | creating a new call independant of existing calls |
| 17:26.48 | docelmo | glogic your being annoying.. present your question and wait for a answer |
| 17:27.06 | glogic | docelmo: i hit the up arrow by accident, relax |
| 17:28.37 | glogic | docelmo: you're in tampa? |
| 17:28.39 | syzygybsd | glogic: it was the whole 5 lines for one question thing. even if we take out the up arrow accident, that is still 4 lines for one question... |
| 17:28.45 | glogic | docelmo: i'm in tampa |
| 17:28.49 | docelmo | uhh if thats what my hostname says |
| 17:28.49 | Hmmhesays | wiki wild wild |
| 17:28.52 | Hmmhesays | wiki wiki wild wild |
| 17:28.57 | glogic | syzygybsd: ok sounds good thanks for nothing |
| 17:29.29 | syzygybsd | wow, that is the first person I have put on ignore |
| 17:29.36 | veepster | ariel:ping |
| 17:29.40 | docelmo | I have done it a few times |
| 17:29.47 | jbroome | in here or on IRC in general? |
| 17:29.55 | syzygybsd | IRC |
| 17:30.00 | jbroome | whoah |
| 17:30.04 | syzygybsd | been on for 6 years too |
| 17:30.20 | jbroome | your tolerence is admirable |
| 17:30.40 | syzygybsd | i normally don't hang out in channels where there are annoying people |
| 17:31.30 | syzygybsd | lol.. he was patient. waited all of 4 minutes for an answer before he left |
| 17:31.47 | Hmmhesays | 4 minutes? |
| 17:31.52 | syzygybsd | maybe less |
| 17:31.59 | *** join/#asterisk glogic (n=gml@66.230.159.3) |
| 17:32.34 | docelmo | glogic to answer your question yes its possible but Im not sure from the cli |
| 17:32.40 | syzygybsd | he isn't in here anymore |
| 17:32.46 | glogic | glogic: i'm right here |
| 17:32.56 | docelmo | syzygybsd you ignored him geesh |
| 17:33.02 | syzygybsd | so he is.. missed the way my clent orders uppercase/lowercase |
| 17:34.36 | jbalcomb | [TK]D-Fender: I changed that setting manually on the phone and it wrote the <mac>-phone.cfg with 'voIpProt.server.1.address' |
| 17:34.42 | *** join/#asterisk Eggplant (i=No@dsl-216-155-213-144.cascadeaccess.com) |
| 17:35.11 | jbalcomb | [TK]D-Fender yeah, but it still doesn't work. haha. |
| 17:35.35 | [TK]D-Fender | jbalcomb : reboot the phoen, then lock it into the boot menu while you wipe that file out. then let it resume and it should get overriden. |
| 17:36.14 | [TK]D-Fender | jbalcomb : I *DID* tell you your use of web-config would come back to haunt you. |
| 17:36.40 | dwrecktion | can someone explain to me exactly what Zaptel means in reference to Asterisk? |
| 17:36.47 | jbalcomb | [TK]D-Fender I've never used the web config on this phone, which is why i beleive it still doesn't work a week after i got it. ;) |
| 17:37.11 | [TK]D-Fender | jbalcomb : because it was never able to write back the "phone" config it was probably keeping it local and always taking that setting over the server. upon upload however I believe it unlocks it and deleteing the file should cause a local wipe |
| 17:37.19 | jbalcomb | dwrecktion: it's for zapping non-voip telephones so that everyone has to buy ip phones |
| 17:37.30 | [TK]D-Fender | jbalcomb : It's never YOUR fault, is it? ;) |
| 17:37.54 | syzygybsd | jbalcomb: that isn't nice... |
| 17:38.09 | jbalcomb | [TK]D-Fender: it's my fault i don't know what i'm doing yes, but it isn't my fault this is a rediculously complicated process. |
| 17:38.31 | [TK]D-Fender | jbalcomb : easy-breezy for me.... |
| 17:38.43 | Idle | I seem to get a LOT of '14' or '20' second messages that are 2 seconds of silence... its like its only recording the silence off the end of the message |
| 17:38.46 | jbalcomb | [TK]D-Fender yeah yeah, thats why i hired you... ;) |
| 17:38.47 | syzygybsd | dwrecktion: zaptel is the interface to PTSN network |
| 17:39.08 | [TK]D-Fender | jbalcomb : Getting closer all the time are we?;) |
| 17:39.32 | jbalcomb | [TK]D-Fender sure seems like it. the DOO breathing down my neck friday was good motivation. |
| 17:39.57 | jbalcomb | [TK]D-Fender this is actually her phone on my desk that i'm trying to get working. |
| 17:40.02 | syzygybsd | DOO? |
| 17:40.20 | syzygybsd | now why was she breathing on your neck.... |
| 17:41.05 | salviadud | yeah, that's kinky dude |
| 17:41.11 | salviadud | i'd hit that |
| 17:41.27 | jbalcomb | syzygybsd because her and an account manager just realized that when no reps are in a queue it plays the announcement twice, puts on music, and then asks them to hold or leave a voicemail |
| 17:41.39 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 17:42.04 | jbalcomb | salviadud shes definitely a MILF but I think she leans towards men her age with money and power |
| 17:42.13 | vooduhal | Hello all. Quick question about the manager API. What is the proper way to read multi-line returns from a command? Ie. Agents |
| 17:42.23 | salviadud | you can fake money and power |
| 17:42.24 | [TK]D-Fender | jbalcomb : "leaveempty=no", "joineempty=yes" |
| 17:42.28 | salviadud | just wear lots of bling |
| 17:42.39 | jbalcomb | salviadud I guess my jar of change and root passwords don't count. :/ |
| 17:42.52 | salviadud | jbalcomb, lol |
| 17:42.57 | *** join/#asterisk [tasty]freeze (n=YamahaBr@pm3.6-4.wctc.net) |
| 17:43.33 | jbalcomb | [TK]D-Fender i definitely don't want that. they want the announcement once, and then straight to voicemail if the queue is unmanned. |
| 17:43.41 | *** join/#asterisk _Sam-- (n=sam@phone2.kneedraggers.com) |
| 17:43.53 | [TK]D-Fender | jbalcomb : When would that happen? |
| 17:44.04 | _Sam-- | hey can anyone recommend any sip clients for PDA/pocket pc? |
| 17:44.10 | [tasty]freeze | I hate to ask, but I just now jumping into the configuration file, and I was trying to find the command to send SIP Headers in the configuration, I have a grandstream gxp-2000 that i want to autoanswer (so its like an intercom). can anyone please send me in the right direction, or help me out here? |
| 17:44.19 | jbalcomb | when our system was built (before my time) the prez requested that we play the announcement before we even put them in the queue. thats why it gets played twice. |
| 17:45.01 | jbalcomb | [tasty]freeze: exten => 14001,5,SIPAddHeader(Call-Info: answer-after=0) |
| 17:45.06 | [TK]D-Fender | jbalcomb : No, I mean when would the queue ever be unmanned? |
| 17:45.10 | *** join/#asterisk ajaymn (i=AJmn@70.59.126.197) |
| 17:45.14 | [tasty]freeze | thank you so much |
| 17:45.17 | [tasty]freeze | jbalcomb |
| 17:45.19 | jbalcomb | [tasty]freeze: that will make the GXP beep and auto answer based on sip info |
| 17:45.23 | jbalcomb | np |
| 17:45.51 | [tasty]freeze | well, one more question, is the second number (5), the order it runs the commands in? becuase that has been my assumption so far. |
| 17:46.01 | jbalcomb | [TK]D-Fender: well, you would think they wouldn't be right? i thought so too. i guess they don't care 100% if there is always someone available. |
| 17:46.01 | *** join/#asterisk enjay- (n=enjay@71.216.165.97) |
| 17:46.24 | jbalcomb | [tasty]freeze: yes, of a sorts. its the priority. |
| 17:46.42 | [TK]D-Fender | jbalcomb : They should let the caller sit in queueu as long as they want and give them the OPTION to leave a VM. |
| 17:46.53 | [tasty]freeze | so wouldn't i want that at the top of the list of things to be sent thru correct? |
| 17:47.03 | [TK]D-Fender | jbalcomb : thats the kind of service I appreciate. |
| 17:47.30 | [TK]D-Fender | jbalcomb : When I want a human you're damned right I'll wait in line, thats what my really nice Polycom speakerphone is for! ;) |
| 17:48.13 | jbalcomb | [TK]D-Fender: i am not a 'business partner' in this company yet. for now i simply accomplish what they want. |
| 17:48.38 | jbalcomb | [tasty]freeze: http://pastebin.ca/90437 |
| 17:48.46 | [tasty]freeze | thank you much |
| 17:49.00 | jbalcomb | [tasty]freeze: quite right. jolly good day fella. |
| 17:49.12 | [tasty]freeze | you too |
| 17:49.37 | jbalcomb | [TK]D-Fender seriously though, i really just need to get this damn polycom to work from the config file via FTP. |
| 17:49.54 | [TK]D-Fender | jbalcomb : while avaoiding "politics", the ability to "seed" them with things like this is to tell them its a great idea and you're glad they brought it up ;) |
| 17:51.08 | jbalcomb | [TK]D-Fender: well, that is certainly there but i'm strangely bad at tact, grace, and consistancy so I've learned to not push instead. |
| 17:51.20 | jbalcomb | for now.... |
| 17:52.05 | CoffeeIV | TripleFFFF: any luck ? I'm about to post this one to the asterisk-users list and move on to the rest of the stuff I have to do today |
| 17:52.49 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 17:54.24 | *** join/#asterisk pdtmobile (n=ptinsley@adsl-070-148-122-203.sip.bna.bellsouth.net) |
| 17:54.26 | [TK]D-Fender | jbalcomb : Oh, and when in doubt do a factory reset on it. |
| 17:54.39 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 17:54.41 | PakiPenguin | hello there |
| 17:54.55 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:55.39 | PakiPenguin | i have a problem , i have an ata registered to asterisk , and a did mapped to it , when a call comes in , it goes thru fine , i pick up and everything is fine , but the call is dropped sometimes after 20 secs , sometimes after 30secs , can anyone tell me the reason? and how can i fix it? |
| 17:55.47 | *** join/#asterisk Paavum (n=asdasdas@200.71.58.39) |
| 17:55.54 | Paavum | Hey |
| 17:55.58 | Paavum | Hello |
| 17:56.11 | Paavum | EEEECHOOOOOO |
| 17:56.11 | Hmmhesays | yup |
| 17:56.12 | jbalcomb | [TK]D-Fender yeah, i've been doing that each time i change something just to feel a 'from the box' feeling. |
| 17:56.21 | syzygybsd | PakiPenguin: do you have an absolutetimeout? |
| 17:56.34 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 17:56.56 | PakiPenguin | nope |
| 17:57.21 | Paavum | Hmmhesays You could at least use a clean sock |
| 17:57.47 | Paavum | Hey is there any way I can recreate a channel that I destroyed with zap destroy channel? |
| 17:58.28 | Paavum | Maybe its the sock... |
| 17:58.41 | Paavum | Hey is there any way I can recreate a channel that I destroyed with zap destroy channel? |
| 17:59.00 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 17:59.09 | Strom_C | Paavum: is there any way you can possibly ask the same question once every second until someone answers it? |
| 17:59.23 | Strom_C | because that would really help everyone, you know |
| 17:59.26 | syzygybsd | Strom_C: don't ask people that, someone will do it |
| 17:59.34 | Paavum | Strom_C, --> No, cuz then I'd get kicked you know |
| 17:59.54 | syzygybsd | only if one of the ops is in here... |
| 18:00.17 | Paavum | They'll be here, eventually |
| 18:00.24 | Paavum | and I'll get kicked, eventually |
| 18:01.41 | syzygybsd | make sure you include all their names on each line, it will make it go faster |
| 18:02.47 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-2-14.nwrknj.east.verizon.net) |
| 18:02.50 | *** join/#asterisk jero (n=jero@OTWAON23-1177993857.sdsl.bell.ca) |
| 18:03.02 | Dovid | mornin again |
| 18:03.18 | syzygybsd | afternoon here |
| 18:03.23 | syzygybsd | by 1 minute |
| 18:04.21 | Paavum | syzygybsd, --> You think so? |
| 18:04.33 | *** part/#asterisk dwrecktion (n=dwreckti@71.16.158.170) |
| 18:04.35 | syzygybsd | try to find out |
| 18:05.33 | Paavum | syzygybsd, --> I've never been so much of a curious guy ... |
| 18:05.35 | PakiPenguin | so? |
| 18:05.36 | Dovid | ur right afternoon |
| 18:05.41 | Dovid | my time is off |
| 18:05.53 | Dovid | i installed asterisk without mpg123 and i dont have any MOH |
| 18:06.03 | Dovid | how do i set it up now using native MOH ? |
| 18:06.23 | *** part/#asterisk jero (n=jero@OTWAON23-1177993857.sdsl.bell.ca) |
| 18:07.20 | jbalcomb | [TK]D-Fender i got it figured out, sort of. I'm going to configure it on the web interface and disect the <mac>-phone.cfg to determine what options i was missing that it needed to work |
| 18:07.26 | syzygybsd | Dovid: http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 18:07.32 | syzygybsd | gives you a couple ways to do it |
| 18:07.55 | jbalcomb | [TK]D-Fender i think part of my struggle is that i didn't think i was going to need to change anything in sip.cfg |
| 18:09.15 | Paavum | Hey, I'm trying to re create a channel I destoyed with destroy channel, how can I do that= |
| 18:09.16 | Paavum | ? |
| 18:09.50 | Strom_C | Paavum: you pick up the phone and redial? |
| 18:11.26 | Paavum | Strom_C, --> The thing is. I dial and they give me a busy tone. when I go to the cli and do a zap show channel xxx |
| 18:11.35 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 18:11.35 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 18:11.43 | Paavum | I see they have owner and they are on-hook |
| 18:11.45 | salviadud | i recentley bought "the book" i think it's a nice read |
| 18:11.50 | salviadud | recently, hehe |
| 18:12.16 | Dovid | if i didnt install mpg123 b4 and i have asterisk running what do i do now to get MOH ? |
| 18:12.23 | salviadud | although, there's lots of functions that have been added since it was published, it's a nice starter kit kinda thing |
| 18:12.30 | Strom_C | Dovid: install mpg123? |
| 18:12.58 | jbalcomb | salviadud: isn't it a bit outdate to be helpful? I felt that as generally telephony book it had credit and maybe on asterisk backend theory too. |
| 18:13.02 | Dovid | do i need to recompile asterisk ? |
| 18:13.19 | jbalcomb | dovid: apt-get install madplay? |
| 18:13.52 | Dovid | jb didnt understand ur last |
| 18:13.53 | jbalcomb | Dovid: it's probably ready to use whatever app you've chosen |
| 18:14.00 | Dovid | i didnt choose anything |
| 18:14.13 | Dovid | just installed asterisk and MOH isnt working |
| 18:14.15 | Dovid | oooooooooooooh |
| 18:14.20 | Dovid | could be timing issue :) |
| 18:14.30 | Dovid | its allways the obvious stuff |
| 18:14.49 | Dovid | im an idiot |
| 18:14.57 | Dovid | i have sangoma. how do i modprobe that ? |
| 18:15.09 | salviadud | jbalcomb, it's not so outdated that everything on it is deprecated. and it's a good reference manual if you forgot a thing or two |
| 18:15.36 | salviadud | jbalcomb, and the intro does a nice bed-time story |
| 18:15.53 | salviadud | jbalcomb, so you can share asterisk with the kids and the wife |
| 18:15.53 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 18:16.06 | Paavum | Hey, I'm trying to re create a channel I destoyed with destroy channel, how can I do that? |
| 18:16.10 | jbalcomb | salviadud true enough sir. ;) |
| 18:16.25 | jbalcomb | Paavum: why would you even think this is possible? |
| 18:17.08 | jbalcomb | Paavum: I'm quite certain you're going to have to reinitiate the call. |
| 18:17.08 | eBody | does anyone know how to configure a tdm2400p, i'm having trouble because of the way it connects to the 66 block punch down. |
| 18:17.38 | Paavum | jbalcomb, --> I have the following issue: |
| 18:17.57 | eBody | will it find any live line that is punched down, or do i have to specify the punches? |
| 18:17.59 | Paavum | a couple of extensions die (no dialtone) |
| 18:18.10 | Paavum | when I ring them I get a busy signal |
| 18:18.33 | Paavum | and when I do a zap show channel |
| 18:18.43 | Paavum | I see they are either in threeway call with themselves |
| 18:18.54 | Paavum | OR they have an owner |
| 18:19.04 | Paavum | butr their state is "onhook" |
| 18:20.54 | rob0 | I'm trying to use the Console as a softphone (chan_alsa.so), but I'm missing something basic. I thought you could just "Dial(IAX2/user:password@host/1234)" to call out. |
| 18:21.45 | eBody | anybody use a 66 block punch down with a tdm2400?? |
| 18:22.01 | docelmo | tdm2400 no.. ampenol yes |
| 18:22.57 | eBody | the 66 block is connected through an amphenol cable |
| 18:23.01 | docelmo | but then again you cant actually punch down a 240 |
| 18:23.06 | docelmo | err tdm2400 |
| 18:23.36 | docelmo | ok so whats the problem? |
| 18:23.41 | eBody | i'm connecting my tdm2400 to this punch down block using an amphenol cable. |
| 18:23.49 | eBody | is this not a correct way of doing this? |
| 18:24.09 | mut | and/or the only way.. |
| 18:25.11 | eBody | my question is how asterisk decides how to name or identify each of the punched down lines. |
| 18:27.12 | Strom_C | eBody: like I told you last time, it follows the standard 25 pair color code |
| 18:28.08 | TripleFFFF | CoffeeIV ? |
| 18:28.10 | Strom_C | eBody: you /do/ know the color code, don't you? :) |
| 18:28.15 | eBody | Strom_C, yeah sorry i left last time. i have it connected correctly. |
| 18:28.29 | eBody | and the phone line also. |
| 18:28.31 | CoffeeIV | TripleFFFF: I'm here |
| 18:28.35 | sevard | syzygybsd: you still around? |
| 18:28.36 | TripleFFFF | roses are red, violets are blue. where all noobs and so are you |
| 18:28.41 | syzygybsd | maybe... |
| 18:28.45 | sevard | shaweet. |
| 18:28.52 | *** join/#asterisk johnnyb (n=jonathan@adsl-38-9-196.tulsaconnect.com) |
| 18:28.59 | eBody | but now what........how do i designate each line coming in. |
| 18:29.07 | Strom_C | what do you mean? |
| 18:29.48 | *** join/#asterisk dwrecktion (n=dwreckti@71.16.158.170) |
| 18:30.01 | johnnyb | Has anyone tried asterisk with three TDM card? Did it work allright? |
| 18:30.04 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 18:30.14 | eBody | my trixbox see my tdm card working properly. and i have the phone line set up also. |
| 18:30.30 | Strom_C | eBody: for trixbox support, go to #freepbx |
| 18:30.36 | eBody | i am just trying to setup a simple dial plan to answer and echo |
| 18:31.44 | dwrecktion | When I pick up the phone I get a dial tone, but as soon as I hit * to try and dial an extension, I get a busy signal. Does this tell me anything in particular, or is it a universal "your system doesn't work" thing? |
| 18:32.13 | Strom_C | dwrecktion: why are you dialing *? |
| 18:32.45 | salviadud | eBody, buddy... i'm sorry |
| 18:33.39 | dwrecktion | Strom_C: that was how our last system was configured and I guess I thought it was the default config? Maybe not? |
| 18:33.52 | Strom_C | dwrecktion: ?? |
| 18:34.20 | dwrecktion | maybe i should be in the trixbox room....hahaha |
| 18:34.26 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 18:34.30 | Strom_C | are you using trixbox? |
| 18:34.35 | dwrecktion | Yes. |
| 18:34.39 | Strom_C | yes, please go there |
| 18:34.44 | [andromeda] | Using an AGI-PHP script, how can I obtain a channel variable? |
| 18:34.50 | dwrecktion | Sorry |
| 18:36.09 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 18:36.53 | *** join/#asterisk johnnyb (n=jonathan@adsl-38-9-196.tulsaconnect.com) |
| 18:37.46 | *** join/#asterisk anthonyl (n=anthony@m815f36d0.tmodns.net) |
| 18:38.27 | SplasPood | heh, Voicepulse is pretty responsive |
| 18:39.13 | *** join/#asterisk andrejkw (n=andrejkw@c-71-57-143-216.hsd1.fl.comcast.net) |
| 18:39.16 | andrejkw | Hey guys |
| 18:39.30 | sevard | when they have low call volume. |
| 18:39.53 | andrejkw | I am having a hard time figuring out how to do Phone -> PAP2 -> Asterisk -> VoipCheap (VoipBuster). |
| 18:40.06 | TrixVox | nah, they're more responsive than every other iax provider any day |
| 18:40.10 | andrejkw | Can someone please help? Any help would be greatly appreciated. |
| 18:40.16 | SplasPood | sevard: heh they sent out some sales email announcing "new lower rates" but it had someone else's account username in it.. So I dropped em an email to let them know.. Got a re-send with the correct info, then a call to confirm shortly thereafter |
| 18:40.43 | SplasPood | Way more than I needed, was just an FYI for them |
| 18:40.47 | TrixVox | every email you send them gets a call back unless you email them first and say the issue is resolved |
| 18:40.58 | SplasPood | hah |
| 18:42.16 | andrejkw | Anyone, please? :'( |
| 18:42.35 | TrixVox | your question is too general |
| 18:42.50 | SplasPood | Voicepulse used to be kinda flaky, but I think they've cleaned up their act quite a bit |
| 18:42.55 | SplasPood | I just use them for inbound at this point.. |
| 18:43.20 | andrejkw | Oh ok, well I need help getting PAP2 working with Asterisk. |
| 18:43.22 | TrixVox | ever since they moved off that piece of crap "gw5.voicepulse.com" it's been rock solid |
| 18:43.29 | *** join/#asterisk gennaro (n=root@ppp-62-10-136-58.dialup.tiscali.it) |
| 18:43.35 | andrejkw | I looked online everywhere, and couldn't find anything on this subject. |
| 18:43.53 | TrixVox | that is very basic functionality and there should be tons of info on the website, anything with a sipura+asterisk will be the same as a pap2 |
| 18:44.13 | TrixVox | i mean "web" not "website" |
| 18:44.49 | andrejkw | Ok, thanks very much ;) |
| 18:44.54 | andrejkw | Found just what I needed. |
| 18:45.02 | sevard | crap |
| 18:45.08 | sevard | indications.c:286 ast_playtones_start: SIP/1338-c46e: tone component 'busy' of 'busy' is no good |
| 18:45.09 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 18:45.13 | r_evolution | ohmigosh. |
| 18:45.14 | r_evolution | heh |
| 18:48.14 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B79EF9.pool.t-online.hu) |
| 18:48.25 | r_evolution | everyone is so quiet... |
| 18:48.28 | r_evolution | it's kinda creepin me out |
| 18:48.40 | file | r_evolution: must be brain slugs |
| 18:49.28 | r_evolution | oooo noez not the brain slugs! |
| 18:50.18 | r_evolution | hey file I thought you guys said 1.4 was going to be 'soon' |
| 18:50.21 | r_evolution | sup witdat? |
| 18:51.03 | TripleFFFF | relative soon is 3 years vs end of life |
| 18:51.21 | TripleFFFF | anyone see how i can edit a 9 gig file in with vi ? |
| 18:51.26 | TripleFFFF | or not enoughr am i got ? |
| 18:51.32 | r_evolution | yeh... soon is definitely a relatively ambiguous concept |
| 18:51.45 | r_evolution | i'm sure you can... but the question is... why would you want to? O_O |
| 18:52.02 | TripleFFFF | edit ? |
| 18:52.07 | TripleFFFF | coz i need to add an index |
| 18:52.09 | r_evolution | edit a 9 GB file yes |
| 18:52.10 | TripleFFFF | to my dump.sql |
| 18:52.14 | r_evolution | ooo |
| 18:52.24 | TripleFFFF | im scared of alter table add index |
| 18:52.36 | r_evolution | i'd be scared of vi dump.sql |
| 18:52.37 | TripleFFFF | well i got a copy now . i eavled the time to 1900 seconds to alter |
| 18:52.38 | r_evolution | but that's just me |
| 18:52.42 | TripleFFFF | since 65k records took 190 |
| 18:52.47 | TripleFFFF | and this is 70 mill |
| 18:52.50 | r_evolution | at least when it's 9 GB |
| 18:53.06 | r_evolution | O_o |
| 18:53.09 | r_evolution | big lotsa CDR? |
| 18:53.14 | TripleFFFF | ';) |
| 18:53.22 | r_evolution | STAY OFF TEH PHONE MANG! |
| 18:53.24 | *** join/#asterisk tenlet (n=tenlet@pool-138-89-84-128.mad.east.verizon.net) |
| 18:53.27 | TripleFFFF | lol |
| 18:53.32 | TrixVox | i wouldn't do add index... |
| 18:53.33 | jbalcomb | [TK]D-Fender: ok, the doc says overrides are stored internally and can ALSO be saved as <mac>-phone.cfg. Deleting the file doesn't get rid of them, you have to 'Reset to Default' -> Local configuration. |
| 18:53.45 | TrixVox | ... on a live database |
| 18:53.54 | r_evolution | pee ess... |
| 18:53.59 | TripleFFFF | here goes nothing |
| 18:54.00 | TripleFFFF | ALTER TABLE `info` ADD INDEX `phone` ( `phone` ) |
| 18:54.02 | TripleFFFF | ;) |
| 18:54.03 | r_evolution | anyone have an effin clue what this means? |
| 18:54.04 | r_evolution | SIP/2.0 415 Unsupported Media Type |
| 18:54.06 | r_evolution | O_o |
| 18:54.08 | TripleFFFF | its my client db |
| 18:54.11 | r_evolution | stupid ATA :( |
| 18:54.18 | TripleFFFF | they just dont know it yet |
| 18:54.23 | r_evolution | O_O |
| 18:54.24 | TrixVox | is it live? all writes will be blocked until it finishes |
| 18:54.30 | TripleFFFF | no problem |
| 18:54.32 | r_evolution | they dont know it's their DB? or they don't know you'll be doing this? |
| 18:54.35 | TripleFFFF | not used for nanything else |
| 18:54.42 | TripleFFFF | j/k |
| 18:54.50 | r_evolution | so you might wanna take the switch down... |
| 18:54.55 | TripleFFFF | not sure i want to that why i made a dump to edit the struct then reimport |
| 18:54.56 | r_evolution | ya know... if they're keeping CDR and all |
| 18:55.00 | TrixVox | and you need enough space for the index left on the db or it'll all go to hell |
| 18:55.00 | TripleFFFF | oh im dump |
| 18:55.17 | TripleFFFF | i need 2 dumpds.. on struct only and other data only.. then i edit struct..drop redo..reimport |
| 18:55.27 | *** join/#asterisk Casterman (n=pierre@kasadsyed.net8.nerim.net) |
| 18:55.34 | jbroome | use colonblow |
| 18:55.36 | TripleFFFF | dib is 8.6 gig got 25 left |
| 18:55.46 | Casterman | hello good evening |
| 18:55.54 | jbalcomb | [TK]D-Fender: adding voIpProt.server.1.address and removing reg.1.server.1.address fixed it. :/ |
| 18:55.57 | TripleFFFF | it does a copy first on alter.. so that means.. 8 left.. index im not sure how much itlls take its an int(11) |
| 18:56.03 | TripleFFFF | so 70 million * 11 |
| 18:56.06 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 18:56.32 | rob0 | Colon Blow? Sounds delicious! |
| 18:57.06 | jbroome | that'll get you all the dumps you need |
| 18:57.08 | sevard | you'd need 40 million bowls of rolled oats to equal just one spoon full of Colon Blow! |
| 18:57.16 | Casterman | i did setup a *@home with a iax trunk how can i check the trunk is working |
| 18:57.17 | Casterman | ? |
| 18:57.39 | jbalcomb | Casterman: I would start by joining the correct channel.. |
| 18:57.39 | sevard | you might want to check a channel that supports asterisk@home |
| 18:57.44 | [TK]D-Fender | jbalcomb : That'd help :) Mind you I never get myself in those messes... did mine right OOB ;) |
| 18:58.04 | jbalcomb | [TK]D-Fender I think I'm going to have you fired. ;) |
| 18:58.33 | jbalcomb | Casterman: In this particular case that'd be #freepbx |
| 18:58.33 | r_evolution | fired out of a cannon? |
| 18:58.36 | r_evolution | that'd be cool. |
| 18:58.54 | jbalcomb | r_evolution: if I ever catch him in the states that might be an option. ;) |
| 18:59.03 | r_evolution | all in favor of firing [TK] out of a cannon... say aye. |
| 18:59.06 | jbalcomb | its only 33 miles across the lake |
| 18:59.07 | r_evolution | yeah i know... |
| 18:59.09 | r_evolution | damn canadians. |
| 18:59.23 | Casterman | <jbalcomb: thx |
| 18:59.38 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 18:59.45 | jbalcomb | Casterman: i do the best i can for lack of a support contract. =) |
| 19:00.10 | *** join/#asterisk watchy (n=gweg@office2.gwhsi.com) |
| 19:00.22 | watchy | whats a program for windows to play .ulaw files? |
| 19:00.35 | Casterman | <jbalcomb: how much you charge for support ? |
| 19:00.35 | jbalcomb | watchy: i think wavepad can do it |
| 19:00.47 | watchy | sweet |
| 19:00.56 | jbalcomb | Casterman: depends on the arrangements but usually 75-150 |
| 19:01.15 | Casterman | E or $ ? |
| 19:01.26 | jbalcomb | USD/hr |
| 19:02.43 | r_evolution | you can also rent strippers for him Casterman |
| 19:03.24 | jbalcomb | r_evolution: that does pretty much cover the difference ;) liquor and cubans are good too. |
| 19:03.58 | r_evolution | you say cubans... are you trying to traffic in the trade of small children again? |
| 19:04.28 | jbalcomb | r_evolution: err.. umm.. no, i was.. ah .. just talking about cigars.. ya know, for smoking. |
| 19:04.39 | watchy | strange |
| 19:04.50 | jbalcomb | indeed |
| 19:04.56 | [TK]D-Fender | :O |
| 19:05.00 | [TK]D-Fender | zomg! |
| 19:05.01 | watchy | wavepad wont play these ulaw files asterisk recorded |
| 19:05.47 | jbalcomb | watchy: that is peculiar. how do you get ulaw files? all my stuff is gsm and wavpad handles those. |
| 19:06.03 | watchy | oh |
| 19:06.09 | *** join/#asterisk Vec (n=Vector@dsl-146-105-111.telkomadsl.co.za) |
| 19:06.10 | watchy | should i have asterisk record in gsm? |
| 19:06.37 | jbalcomb | watchy: i don't know whether one is better than the other for any reason |
| 19:06.46 | watchy | but you use gsm? |
| 19:07.04 | jbalcomb | watchy: yeah, it was already set up when i got here though |
| 19:08.58 | *** join/#asterisk anonymouz666 (n=anonymou@20151155235.user.veloxzone.com.br) |
| 19:09.11 | r_evolution | yeah i usually go for the gsm as well :) |
| 19:09.20 | r_evolution | and quicktime will play those |
| 19:10.27 | Vec | I have a Sipura 3000, and am trying to use it as a SIP > PSTN gateway, exten => 0,1,Dial(SIP/sipura2@sipura2,10,D(5552835)), works but then then it will always dial that number, how can I set it up, so it takes the digits from the dialed number ? |
| 19:10.35 | r_evolution | what an effin cunt. |
| 19:11.06 | watchy | sweet |
| 19:11.23 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B79EF9.pool.t-online.hu) |
| 19:16.01 | *** join/#asterisk droops (n=root@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 19:16.38 | rob0 | Nutsy idea ... but ... is PCMCIA-based zaptel hardware available? (Not finding anything in Google.) |
| 19:17.30 | r_evolution | O_o |
| 19:17.41 | jbalcomb | [TK]D-Fender: ok, brand new phone, OOB, booting directly from precreated config files via FTP.... |
| 19:17.44 | r_evolution | interesting concept at least :) |
| 19:18.43 | jbalcomb | rob0 isn't it called PCCard now? |
| 19:19.19 | rob0 | Yeah, I figure a portable PBX could be had by means of connecting ATA's. (And it's always PCMCIA to me. :) ) |
| 19:19.55 | jbalcomb | rob0 seems like its sometimes called CardBus as well |
| 19:20.00 | [TK]D-Fender | jbalcomb : upgrade and provision in progress? |
| 19:20.20 | jbalcomb | [TK]D-Fender updagraded and provisioned!! just tested it!! woot!! |
| 19:22.33 | TripleFFFF | trxtel here ? |
| 19:24.00 | *** join/#asterisk effectiveape (n=nick@82.153.22.16) |
| 19:24.05 | *** join/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 19:24.26 | sevard | Can you set astdb variables in sip.conf? |
| 19:24.29 | effectiveape | *sigh* still having problems :/ |
| 19:24.43 | sevard | because that would be nifty. |
| 19:25.45 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 19:25.47 | sevard | sweet, you can! |
| 19:26.49 | effectiveape | right. I've managed to get the incomming isdn cooking but now there seems to be a problem with my extensions.conf. I have an extension 10 which (at the moment) i just have in the context which i've put the icomming isdn in. However it keeps (http://pastebin.ca/90516) |
| 19:26.55 | [TK]D-Fender | jbalcomb : w00t |
| 19:27.02 | effectiveape | But when i call the same extension from another internally it's fine |
| 19:27.22 | jbalcomb | sevard: yes |
| 19:27.37 | *** part/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 19:29.29 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:30.02 | effectiveape | my extensions conf (snippet) is http://pastebin.ca/90519 |
| 19:30.42 | effectiveape | and zapata is http://pastebin.ca/90520 |
| 19:30.56 | effectiveape | i'm sure it's something obvious but i can't figure it :/ |
| 19:31.06 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.231.225) |
| 19:31.17 | TripleFFFF | CoffeeIV ? |
| 19:31.18 | Strom_C | effectiveape: is your DNIS on a six digit basis from the telco? |
| 19:31.43 | effectiveape | that's what appears to be comming in. |
| 19:31.55 | [TK]D-Fender | sevard : Astdb in sip.conf? news to me... channel variables sure.... |
| 19:31.58 | stoffell_h | effectiveape: and is the extension 10 forwarded to 99 ?? |
| 19:32.11 | sevard | [TK]D-Fender: example: astdb=chan2ext/SIP/grandstream1=1234 ; ensures an astDB entry exists |
| 19:32.12 | effectiveape | no. I'm not sure where that's happening. |
| 19:32.12 | Strom_C | also why do you have two "1" priorities? |
| 19:32.25 | [TK]D-Fender | sevard : wow, never notived |
| 19:32.26 | effectiveape | It's meant to just goto the dial macro |
| 19:32.29 | sevard | [TK]D-Fender: pretty nifty |
| 19:33.25 | [TK]D-Fender | sevard : Does it override the value on a new channel every time or only if blank? |
| 19:33.43 | Strom_C | effectiveape: why does your extension have two lines with priority "1" |
| 19:34.13 | effectiveape | mistake ;) - i've changed it |
| 19:34.21 | effectiveape | no change on the issue however ;) |
| 19:34.39 | CoffeeIV | TripleFFFF: here |
| 19:34.44 | Strom_C | comment those lines out and change it to exten => _X. and see what the telco sends you |
| 19:35.02 | sevard | [TK]D-Fender: I don't know |
| 19:36.11 | effectiveape | Strom_C: Extension '642042' in context 'inbound-from-isdn' from '' does not exist. Rejecting call on channel 0/1, span 1 |
| 19:36.24 | effectiveape | Strom_C: with exten => _X,1,Macro(stdexten,10,${KERRY}) |
| 19:36.29 | Strom_C | no |
| 19:36.31 | Strom_C | I said _X. |
| 19:36.33 | Strom_C | not _X |
| 19:36.40 | effectiveape | lol cool ;) |
| 19:36.44 | sevard | lolololololo |
| 19:36.51 | sevard | :-| |
| 19:37.04 | effectiveape | hey we all start somewhere ;) |
| 19:37.35 | effectiveape | exactly the same problem |
| 19:37.39 | Strom_C | ok, im off |
| 19:38.49 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) |
| 19:38.50 | [TK]D-Fender | effectiveape : pastebin your extensions.conf and the CLI output of the failed call |
| 19:39.15 | effectiveape | k |
| 19:39.32 | effectiveape | failure: http://pastebin.ca/90516 |
| 19:40.10 | [TK]D-Fender | effectiveape : thats an OLD error message, doa NEW call please |
| 19:40.18 | effectiveape | ok |
| 19:40.38 | *** join/#asterisk cHr1Zt1An (n=cHr1Zt1A@200.121.195.177) |
| 19:41.26 | *** join/#asterisk chorlick (n=chorlick@gateway.digium.com) |
| 19:41.51 | effectiveape | http://pastebin.ca/90529 |
| 19:42.42 | effectiveape | http://pastebin.ca/90532 error |
| 19:43.41 | [TK]D-Fender | effectiveape : Ok, so your phone is force forwarding to 99 and it doesn't exist... |
| 19:43.48 | stoffell_h | effectiveape: who is 10.0.2.141 ? |
| 19:43.49 | [TK]D-Fender | effectiveape : Seems pretty straightforward. |
| 19:43.56 | effectiveape | that's the ata |
| 19:44.21 | stoffell_h | effectiveape: well, as [TK]D-Fender says, it's forwarded.. |
| 19:44.27 | effectiveape | but i don't understand why it's doing that since when i'm calling extension -> extension its all fine but when i'm calling in from external its not |
| 19:44.44 | [TK]D-Fender | effectiveape : pastebin and internal call to it then. |
| 19:44.47 | [TK]D-Fender | an* |
| 19:45.14 | [TK]D-Fender | effectiveape : it could be a SELECTIVE forwarding.... hard to say. |
| 19:45.47 | enjay- | If a call comes in over a PRI, how can you route that to an IVR on a seperate server via an IAX trunk? |
| 19:45.55 | effectiveape | http://pastebin.ca/90538 |
| 19:46.23 | effectiveape | where would that forwarding be happening? |
| 19:46.54 | [TK]D-Fender | effectiveape : must be a selective forwarding then..... and on the ATA itself |
| 19:50.04 | *** join/#asterisk jm|home (n=jamiem@dsl-217-155-242-137.zen.co.uk) |
| 19:51.28 | *** join/#asterisk Iam8up|lpy (n=iam8up@cpe-24-210-253-66.woh.res.rr.com) |
| 19:51.36 | Iam8up|lpy | anyone know the number to contact digium for RMAing? |
| 19:51.46 | Iam8up|lpy | on their website they've got a shiny 404... http://www.digium.com/en/products/returnpolicy.php |
| 19:51.59 | enjay- | just call support |
| 19:51.59 | docelmo | uhh how about contactus? |
| 19:52.05 | Corydon-w | Just call the main number and talk to support |
| 19:52.13 | *** join/#asterisk veepster_ (n=veepster@67.130.38.2) |
| 19:52.20 | enjay- | 256.428.6000 |
| 19:52.23 | file | Iam8up|lpy: that page does work for me, and as everyone has said... call support |
| 19:52.37 | file | ah |
| 19:52.39 | file | the contact link off it |
| 19:52.46 | Iam8up|lpy | file - at the bottom, the link is broken |
| 19:53.14 | Iam8up|lpy | oh well, thakns anyways |
| 19:53.41 | file | RMA is 256-428-6270 |
| 19:53.56 | Iam8up|lpy | ty |
| 19:54.17 | Iam8up|lpy | funny how i know where that phone number is located... |
| 19:54.24 | Iam8up|lpy | 256 is huntville - i was there last week =) |
| 19:54.41 | file | crazy that, so was I! |
| 19:54.46 | Iam8up|lpy | neat =) |
| 19:54.48 | Iam8up|lpy | where did you go? |
| 19:54.53 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 19:55.09 | file | somewhere... |
| 19:55.17 | Iam8up|lpy | lol... |
| 20:02.25 | veepster_ | hey, I set up asterisk last week. It works fine internally, but when I try to dial out I keep getting 'all circuits are busy' |
| 20:02.42 | veepster_ | I have trunked it with an analog line using a spa-3000 modem |
| 20:02.50 | veepster_ | is this a common troubleshooting issue..? |
| 20:02.58 | r_evolution | hmmmm |
| 20:03.07 | *** join/#asterisk chorlick (n=chorlick@gateway.digium.com) |
| 20:03.08 | r_evolution | file? in huntsville?... can't seem to think of why O_o |
| 20:03.08 | effectiveape | flipin' ata! |
| 20:03.17 | effectiveape | Was a default setting in there :/ |
| 20:03.43 | effectiveape | Don't know what was triggering it but it's not there anymore - thanks! :) |
| 20:03.53 | Casterman | can i chanel video thru iax ? or sip ? |
| 20:04.28 | file | I was at the Digium office omg |
| 20:04.42 | r_evolution | NO WAY! |
| 20:04.45 | r_evolution | surely not! |
| 20:04.48 | r_evolution | ;) |
| 20:05.22 | r_evolution | next thing you know... you're gonna be telling me that kram fellow was there too! |
| 20:05.47 | file | I know, it's crazy |
| 20:05.54 | r_evolution | definitely... |
| 20:06.01 | rob0 | I'm unclear on the use of regexten in iax/sip.conf. Should that be the name of the [section] in the iax/sip.conf on the other end? Or an extension in the dialplan? Or both? |
| 20:06.34 | file | rob0: it's an extension in the dialplan, when the peer is registered an entry is inserted at priority 1 with a Noop that makes the extension exist essentially |
| 20:07.12 | rob0 | in whatever context was specified, I guess. |
| 20:07.22 | r_evolution | i'm unclear as to why this company wants a man who doesnt even enunciate enough to say 'ACCELERATOR' to be the one to record the voice prompts |
| 20:07.36 | r_evolution | instead of "accelerator" it come's out "celerator" |
| 20:07.39 | r_evolution | comes* |
| 20:08.40 | sevard | there needs to be an 'extensions reload | less' |
| 20:09.08 | r_evolution | hell man there just needs to be less |
| 20:09.11 | *** join/#asterisk burnproof (n=burnproo@210.213.242.145) |
| 20:09.18 | rob0 | If your console is in screen, there's ^A-[ to scroll up. |
| 20:09.19 | sevard | that'd be tight. |
| 20:09.27 | sevard | i always use screen |
| 20:09.39 | rob0 | (PgUp/PgDn should work in copy mode.) |
| 20:09.44 | sevard | i just don't want my screen buffer to be 1000 pages. |
| 20:09.52 | rob0 | (Depending on your terminal, of course.) |
| 20:10.07 | sevard | your screens buffer is different than your terminals line buffer |
| 20:10.32 | r_evolution | O_O |
| 20:10.44 | sevard | 0_p |
| 20:10.47 | r_evolution | i guess it'd fold you up into thirds? |
| 20:10.49 | r_evolution | ouch. |
| 20:11.27 | sevard | asterisk calls my phone every 2 hours from 9am to 6pm and tells me to move my car. |
| 20:11.53 | r_evolution | sounds like the Nag function on wife 1.0 |
| 20:12.05 | sevard | that's actually a downgrade from girlfriend .75 |
| 20:12.23 | effectiveape | more noob anoyances for you from me ;) |
| 20:12.24 | r_evolution | yeah... Mistress 9.0 :) |
| 20:12.38 | h3x0r | > unload chan_flaatio.so |
| 20:12.39 | r_evolution | LUCKILY for me... i dont plan to install wife 1.0 |
| 20:12.52 | sevard | I heard wife 1.0 is actually a trojan |
| 20:12.58 | effectiveape | Executing Dial("SIP/10-7abf", "Zap/g1/07970xxxxxx") in new stack |
| 20:13.06 | r_evolution | for what? mini-me? |
| 20:13.29 | sevard | rob0 married one on the deathbed |
| 20:13.30 | effectiveape | The number displayed is correct however i'm getting a BT 'number not recognised' woman |
| 20:13.50 | effectiveape | Do you have to send a full msisdn? ie(447?) |
| 20:13.58 | effectiveape | for isdn2e |
| 20:14.06 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 20:14.12 | sevard | r_evolution: backup systems are good, especially when they're younger with more options |
| 20:14.24 | r_evolution | i cant get too much younger sevard :) |
| 20:14.30 | sevard | r_evolution: sick. |
| 20:14.32 | r_evolution | current = 22... back-up = 18 |
| 20:14.36 | Casterman | anyone knows about videoconferencing and the iax protocol |
| 20:14.46 | sevard | current 19, backups all 18 |
| 20:14.46 | r_evolution | other back-up = 20 |
| 20:14.48 | r_evolution | soooooooo |
| 20:15.07 | r_evolution | i dunno... the last was one of those 'barely legal' types |
| 20:15.12 | r_evolution | sorta like the current back-up |
| 20:15.20 | r_evolution | and i swear i've never come so close to choking a bitch |
| 20:15.34 | sevard | times when you have to take out the coat hanger |
| 20:15.54 | TripleFFFF | chan_sip.c:9735 handle_response_register: Failed to authenticate on REGISTER to ' |
| 20:15.59 | r_evolution | PIMP STICK! |
| 20:16.02 | TripleFFFF | is there a way to make sur this works ? |
| 20:16.07 | TripleFFFF | like max_Register=999999999999999 |
| 20:16.11 | r_evolution | i already snapped the current on the ass with a belt in the middle of wal-mart |
| 20:16.11 | TripleFFFF | or something |
| 20:16.16 | r_evolution | i got punched in the chest for that :( |
| 20:16.29 | sevard | mine takes it and loves it. |
| 20:17.02 | r_evolution | yeah mine isn't exactly fond of the whole public embarrasment thing |
| 20:17.16 | hi365 | Hi! im getting ZT_CHANCONFIG failed on channel 3: Invalid argument (22) from ztcfg -vv here is error and files: http://pastebin.ca/90565 |
| 20:17.17 | r_evolution | which is why i insist on molesting her in public as often as possible :) |
| 20:17.19 | r_evolution | i call that funny :) |
| 20:17.32 | sevard | ^5 |
| 20:18.25 | r_evolution | :-D |
| 20:18.43 | r_evolution | i flashed a waiter with my ex-gf's ass... in the middle of a japanese steak house... ;) |
| 20:19.07 | r_evolution | i told her thats what she gets for wearing something that's going to cause me dirty thoughts... |
| 20:19.23 | burnproof | hi365: what FXO/FXS do you use? |
| 20:19.24 | effectiveape | anyone? :) |
| 20:19.31 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:19.54 | hi365 | <burnproof> sangoma a200 |
| 20:21.06 | burnproof | oic, could you tail -f /var/log/messages |
| 20:21.19 | *** join/#asterisk gmfm (i=gmfm@rtr.enterprisemtg.net) |
| 20:21.25 | burnproof | if you get the error, invalid/obsolete firmware you are getting a beta test card |
| 20:22.14 | *** join/#asterisk DasTech (n=DasTech@c-67-176-28-65.hsd1.co.comcast.net) |
| 20:22.28 | burnproof | hi365: wanrouter start did you get the error invalid or obsolete/firmware ? |
| 20:22.45 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 20:23.25 | hi365 | http://pastebin.ca/90573 |
| 20:23.39 | sevard | Yoko Kanno - Tank! (Luke Vibert Remix) |
| 20:23.43 | sevard | that's a pretty sweet remix |
| 20:23.48 | burnproof | hi365: as i told you could you please try to load wanrouter start then check your /var/log/messages |
| 20:24.10 | burnproof | hi365: w8 |
| 20:24.16 | burnproof | hi365: let me check |
| 20:24.22 | r_evolution | what in the hello? |
| 20:24.25 | r_evolution | hell* |
| 20:24.36 | r_evolution | Yoko Kanno? wtf is that? j-pop? |
| 20:24.54 | burnproof | hi365: did you compile it with hardware echo cancellation support? |
| 20:25.02 | hi365 | no dont have that |
| 20:25.13 | sevard | yoko kanno wrote all the bebop for cowboy bebop, the only anime that is actually good |
| 20:25.16 | burnproof | hi365: but did you enable HDLC stuff in it? |
| 20:25.20 | sevard | she writes good shit |
| 20:25.33 | burnproof | sevard: i agree with you cowboy bebop is great :) |
| 20:25.35 | hi365 | HDLC? |
| 20:25.50 | sevard | burnproof: word dude, i absolutly hate anime but i love cowboy bebop. |
| 20:26.00 | burnproof | yup upon installation of wanpipe drivers? |
| 20:27.08 | r_evolution | AHHHHHHHH |
| 20:27.14 | r_evolution | i like anime :) |
| 20:27.19 | r_evolution | and i like cowboy bebop |
| 20:27.26 | r_evolution | but i still dont like j-pop :) |
| 20:27.34 | burnproof | :) |
| 20:27.38 | r_evolution | i like to understand what the fuck im listening... assuming it has lyrics... |
| 20:27.46 | r_evolution | in the case of EDM... which is just comprised of vocal samples |
| 20:27.53 | r_evolution | i can deal with not understandin the lyrics :) |
| 20:27.54 | hi365 | <burnproof> whats HDLC |
| 20:27.57 | hi365 | ? |
| 20:28.05 | sevard | i hate jpop too dude, yoko kanno isn't jpop :) |
| 20:28.23 | sevard | i like jazz, tones are universal. |
| 20:28.34 | r_evolution | well i have teh bebop movie... ill go listen to the intro when i get home |
| 20:28.35 | r_evolution | ahhhh |
| 20:28.37 | r_evolution | yeah i did like that |
| 20:29.57 | sevard | soma.fm's spy channel is pretty good. |
| 20:30.24 | burnproof | hi365: ok, to simply it what version of wanpipe did you get from them? |
| 20:31.02 | rob0 | "res_musiconhold.c:511 monmp3thread: Request to schedule in the past?!?!" I'm running ntpd on here, and synch'ed. What else could I check about that? |
| 20:31.25 | hi365 | burnproof: beta4-2.3.4 |
| 20:31.37 | *** join/#asterisk topping (n=topping@adsl-68-122-71-30.dsl.pltn13.pacbell.net) |
| 20:31.53 | *** join/#asterisk Trazz (n=traderz@65.114.86.29) |
| 20:32.11 | burnproof | hi365: we are using the same version, did you try to recompile it? or if you will allow me, can you give access on the machine so that i can check? |
| 20:32.24 | *** part/#asterisk Uboo (n=traderz@65.114.86.29) |
| 20:33.20 | sevard | Is there an asterisk function/command out there that will substiute? we have REGEX but that just matches. |
| 20:34.37 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-138-106.red.bezeqint.net) |
| 20:36.31 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 20:38.42 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 20:42.10 | benjk | sevard, substitute what? |
| 20:45.07 | MikeJ[Laptop] | not nice |
| 20:45.27 | Hmmhesays | Never claimed to be all that nice |
| 20:46.30 | benjk | why does the caller ID from US callers show up as 10 digits only if I am not in the US? Go figure |
| 20:48.08 | MikeJ[Laptop] | Hmmhesays, nice |
| 20:50.22 | Hmmhesays | MikeJ[Laptop]: you ever work on as5300's? |
| 20:53.27 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 20:53.33 | *** join/#asterisk ChkDigit (n=mike@static24-72-137-23.regina.accesscomm.ca) |
| 20:54.06 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001217014b92.vc.shawcable.net) |
| 20:55.27 | Dr-Linux | dlynes_laptop: is back :) |
| 20:56.59 | dlynes_laptop | Dr-Linux, hey |
| 20:57.11 | dlynes_laptop | Just been super busy the last week or so |
| 20:57.25 | dlynes_laptop | still super busy...just not super stupendously busy now |
| 20:57.41 | dlynes_laptop | working on two projects, plus my main job |
| 20:57.50 | sevard | hey dlynes |
| 20:57.54 | sevard | long time no see |
| 20:57.58 | harryvv | interesting aah has been bought out by trixbox |
| 21:00.08 | sevard | http://www.voip-info.org/wiki-Asterisk+config+sip.conf says astdb=chan2ext/SIP/grandstream1=1234 |
| 21:00.13 | drray | bought out? |
| 21:00.17 | sevard | near as i can tell, you can't have a / in there |
| 21:00.31 | sevard | also, near as i can tell, that doesn't actually work at all |
| 21:01.20 | *** join/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk) |
| 21:01.44 | burnproof | has anyone from you guys, successfully compile zaptel-1.2.6 on CentOS release 4.3 |
| 21:01.56 | sevard | i changed it to chan2ext/SIP-grandstream1=1234 (following the example) and DB_EXISTS returns 0 |
| 21:01.59 | burnproof | i presume it's a stock kernel on centos hi365 |
| 21:02.39 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 21:02.51 | carl0s- | hmm. does anyone know what "Calling (out INV)" means on the Cisco phones (7960)? I just SIP'd this one (8.2) and my first call seemed fine. I stopped the call from the other end (mobile phone.. hit busy) and now it won't call out. It's just silent. I can't even call my own extension, it just sits there silently. |
| 21:03.55 | sevard | so i guess my question there is, can anyone see (from the stated data) what i might be doing wrong |
| 21:06.46 | carl0s- | seems I'm not the only one who's had this "out INV" thing |
| 21:08.08 | [andromeda] | Does anyone here know anything about the AGI command while ($env=read()) { |
| 21:08.08 | [andromeda] | $env = str_replace("\"","",$env); |
| 21:08.08 | [andromeda] | $s = split(": ",$env); |
| 21:08.08 | [andromeda] | $agi[str_replace("agi_","",$s[0])] = trim($s[1]); |
| 21:08.08 | [andromeda] | if (($env == "") || ($env == "\n")) { |
| 21:08.10 | [andromeda] | break; |
| 21:08.12 | [andromeda] | } |
| 21:08.16 | [andromeda] | ack, sorry |
| 21:08.43 | [andromeda] | Does anyone here know anything about the AGI command: GET VARIABLE |
| 21:10.17 | dlynes_laptop | heya sevard |
| 21:10.24 | dlynes_laptop | sup, bitchcakes? |
| 21:10.24 | *** part/#asterisk DasTech (n=DasTech@c-67-176-28-65.hsd1.co.comcast.net) |
| 21:10.44 | sevard | having issues with sip.conf meeting astdb |
| 21:10.49 | carl0s- | How bizzare. I've just done "zap show channels" whilst trying to call into my * box through the landline on the TDM400P, and the asterisk console has frozen. (asterisk -r). |
| 21:11.16 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
| 21:11.29 | sevard | carl0s-: disconnect, reconnect, issue the command again |
| 21:11.44 | dlynes_laptop | sip.conf meeting astdb? |
| 21:11.53 | sevard | http://www.voip-info.org/wiki-Asterisk+config+sip.conf says astdb=chan2ext/SIP/grandstream1=1234 |
| 21:11.54 | dlynes_laptop | mind repeating that in English that makes sense? |
| 21:12.06 | carl0s- | sevard: I think * still thinks I'm on the call to my mobile phone (the call that I refused on my mobile.) How can I check this? |
| 21:12.16 | sevard | near as i can tell, your key can't contain a slash |
| 21:12.25 | sevard | near as i can tell, that line doesn't work anyway |
| 21:12.33 | dlynes_laptop | sevard, the key separator is a slash |
| 21:12.48 | sevard | yes, and that's still what voip-info says |
| 21:12.58 | dlynes_laptop | sevard, that's how astdb determines which branch your db node is on |
| 21:13.01 | sevard | i changed it to - and DB_EXISTS still says 0 |
| 21:13.08 | dlynes_laptop | sevard, the '/' separates branches |
| 21:13.23 | carl0s- | OK. zap show channels is returning nothing. zilch. |
| 21:13.29 | sevard | chan2ext/SIP-grandstream1 says 0 (changed in extensions.conf and sip.conf) |
| 21:13.49 | dlynes_laptop | carl0s-, you didn't run ztcfg -vvvvvvvvvvv, and/or you don't have zapata.conf correctly configured |
| 21:14.00 | sevard | reloaded sip and extensions |
| 21:14.03 | dlynes_laptop | carl0s-, and/or you don't have chan_zap.so loaded |
| 21:14.17 | tzafrir_laptop | ztcfg -vv can be misleading |
| 21:14.23 | sevard | if he didn't have chan_zap loaded, zap show channels woudl spit invalid command |
| 21:14.39 | tzafrir_laptop | if it returns an error it man not have actually configured that channel |
| 21:14.58 | carl0s- | dlynes_laptop: well, I used genzaptelconfig. It was working a minute ago. It showed pseudo and 4 (the single x100m on my tdm400). Now when I do zap show channels, I get nothing. calls aren't working on my Cisco phone, and when I follow with "?" for help on a blank line, the asterisk console freezes. |
| 21:15.48 | tzafrir_laptop | what asterisk version? what distro? |
| 21:15.51 | carl0s- | zap show status works. It says OK for my tdm400 and unconfigured for ztdummy. |
| 21:16.01 | dlynes_laptop | carl0s-, have you issued a stop command at all? |
| 21:16.12 | carl0s- | dlynes_laptop: no. should I? |
| 21:16.14 | dlynes_laptop | carl0s-, i.e. a stop command that hasn't completed yet? |
| 21:16.19 | carl0s- | no |
| 21:16.50 | carl0s- | tzafrir_laptop: 1.2.9.1 on Centos erm 4.3 |
| 21:16.54 | dlynes_laptop | ok, just asking because i've seen that behaviour happen after a stop command's been executed, but hasn't taken effect yet |
| 21:17.29 | carl0s- | I have that silly Flash Operator Panel here, and that seems to think my extension is still connect to the cellphone number I called to test a minute ago. |
| 21:18.34 | tzafrir_laptop | carl0s-, is there still a chnnel? show channels |
| 21:18.44 | ChkDigit | unit3 - Audio Warehouse sells iRiver players that play ogg. |
| 21:18.54 | ChkDigit | ... Wrong channel... |
| 21:18.57 | ChkDigit | =) |
| 21:19.30 | carl0s- | tzafrir_laptop: zap show channels is returning nothing. zilch. it's as though ZAP has died on me, but also when I call my own extension from the Cisco 7960, I don't hear a thing. No "this extension is busy" or anything |
| 21:19.42 | dlynes_laptop | carl0s-, do you have disconnect supervision on that line? |
| 21:20.14 | dlynes_laptop | carl0s-, also, what does dmesg say about your zaptel configuration? |
| 21:20.42 | carl0s- | dlynes_laptop: huh? :D If that's something from the telco, then I'd have to say no. I've never heard of it and this is a plain ordinary British Telecom POTS line. I check out the dmesg and there were no problems coming up recently. No funny output from zap or anything. I'll have another look now. |
| 21:21.08 | [andromeda] | Does anyone here know anything about the AGI command: GET VARIABLE? When I try to use this command, and then try to read output from AGI, I get an empty string. However, I ran debug in the Asterisk CLI, and can clearly see Asterisk sending output after the GET VARIABLE command. |
| 21:21.09 | dlynes_laptop | carl0s-, yeah...that's something from the telco |
| 21:21.31 | carl0s- | Can I kill/restart zap easily? without rebooting the whole machine. Perhaps I should upgrade to the new zap 1.2.7 (i'm on 1.2.6) |
| 21:21.39 | dlynes_laptop | carl0s-, disconnect supervision means the telco will switch polarity on tip/ring on hangup, to signal the hangup |
| 21:21.47 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 21:21.57 | hads | Or drop battery. |
| 21:22.02 | dlynes_laptop | carl0s-, restart when convenient (that'll make sure nobody's making a call before restarting asterisk) |
| 21:22.21 | dlynes_laptop | carl0s-, 1.2.6 is fine |
| 21:22.29 | dlynes_laptop | carl0s-, a lot of us have been running it for a while now |
| 21:22.32 | carl0s- | dlynes_laptop: ah. I see. That stops the usual thing where the calling party keeps the line up for ever until they hang up. I think over here it used to work like that - the called party couldn't end the call. I think it might have changed now though. |
| 21:22.52 | carl0s- | dlynes_laptop: sorry, that was a question. I missed the ? |
| 21:23.07 | dlynes_laptop | carl0s-, well, anyways...call up BT, and ask them if they have disconnect supervision enabled on your analog line |
| 21:23.15 | dlynes_laptop | carl0s-, if they don't, ask them to enable it |
| 21:23.26 | carl0s- | dlynes_laptop: ok. |
| 21:23.30 | rob0 | Quoth BT: "huh?" |
| 21:23.34 | carl0s- | :D |
| 21:23.40 | carl0s- | rob0: my thoughts exactly |
| 21:23.59 | carl0s- | dlynes_laptop: "restart when convenient". is that a command, or a suggestion to me? |
| 21:24.07 | hads | It's a CLI command |
| 21:24.12 | carl0s- | oh ok. |
| 21:24.14 | carl0s- | thanks |
| 21:24.37 | carl0s- | oops. that caused the CLI to die. It's frozen now. |
| 21:24.45 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 21:24.58 | dlynes_laptop | carl0s-, beautiful |
| 21:24.58 | rob0 | Perhaps it's running on a different PID. |
| 21:25.06 | dlynes_laptop | carl0s-, seems like you've got other issues |
| 21:25.18 | hads | segfault? |
| 21:25.31 | carl0s- | checking /var/log/asterisk/full now |
| 21:25.41 | dlynes_laptop | carl0s-, are you using mpg123? |
| 21:26.18 | carl0s- | dlynes_laptop: not as far as I know. the log doesn't show anything for the last half hour. |
| 21:26.34 | dlynes_laptop | carl0s-, KILL, KILL, KILL THE EVIL SCOURGE OF MPG123 RIGHT NOW!!!!!! Kill it!!! Kill it!!! Kill it!!!! |
| 21:26.44 | sevard | psycho. |
| 21:26.48 | dlynes_laptop | carl0s-, type ps auxffww | grep mpg123 |
| 21:26.54 | hads | heh |
| 21:27.09 | hads | Huh? full doesn't show anything? THat doesn't sound right. |
| 21:27.12 | carl0s- | nope. no mpg123 here :) |
| 21:27.18 | dlynes_laptop | It's beyond me why it's the default mp3 player for asterisk |
| 21:27.24 | dlynes_laptop | It's absolutely horrible |
| 21:27.32 | hads | Not anymore though. |
| 21:27.38 | tzafrir_laptop | dlynes_laptop, 'ff' in the ps command? |
| 21:27.42 | dlynes_laptop | native is the default now? |
| 21:27.55 | carl0s- | hads: well. it shows nothing since when it all started to go wrong about half an hour ago, when I dialled my mobile from the 7960, I pressed "go away" on my mobile (cellphone to you americans), and it all went wrong. |
| 21:27.56 | dlynes_laptop | tzafrir, ff == full, ww = wide |
| 21:28.05 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 21:28.06 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcd.mn.charter.com) |
| 21:28.14 | hads | Um, I don't know about 1.2 but in trunk native is default. |
| 21:28.33 | hads | carl0s-: I'm not in the US, I know what a mobile is :) |
| 21:28.55 | carl0s- | gonna reboot the machine and see what happens. I had issues where asterisk would segfault followed by a kernel panic with the X100P in there. |
| 21:29.00 | carl0s- | hads: oh good :D |
| 21:29.23 | hads | Are you at the console or remote? |
| 21:29.48 | carl0s- | hads: both. I've got two ssh sessions open and a KVM here. I switched over on the KVM and ran asterisk -r and the CLI froze just the same. |
| 21:29.53 | dlynes_laptop | carl0s-, you're using an x100p for your zaptel channel? |
| 21:30.02 | dlynes_laptop | carl0s-, there's your problem right there |
| 21:30.07 | hads | Ug. |
| 21:30.13 | carl0s- | dlynes_laptop: no I replaced it with a TDM400P single FXO (tdm01b?) just before. |
| 21:30.21 | hads | Yay. |
| 21:30.30 | dlynes_laptop | ok...marginally better |
| 21:30.32 | carl0s- | :D |
| 21:30.44 | hads | carl0s-: Can you ssh in another session and kill asterisk? |
| 21:30.57 | hads | dlynes_laptop is a Sangoma lover :) |
| 21:31.06 | dlynes_laptop | Well, yeah |
| 21:31.23 | carl0s- | well, I have a GSM <-> SIP gateway on its way from Taiwan too. that'll be fun. combined with a second Orange mobile contract which gives free inter-company calls. |
| 21:31.23 | dlynes_laptop | they're nice and stable |
| 21:31.26 | dlynes_laptop | zero issues with line quality |
| 21:31.46 | *** join/#asterisk Vec (n=Vector@dsl-146-105-111.telkomadsl.co.za) |
| 21:31.47 | hads | I will be having a play with them at some stage. Havent got spare coin at the moment. How do you find the drivers to work with? |
| 21:31.55 | carl0s- | I was considering the A200 and the price was similar. Ho hum. |
| 21:32.07 | Vec | I think someone just spoke to me, but I missed it ? |
| 21:32.08 | Mercestes | emerge -av wanpipe if your in gentoo....:) |
| 21:32.10 | hi365 | carl0s- where di u find a sip-gsm gateway? |
| 21:32.13 | dlynes_laptop | hads, the drivers since i started working with them seem to be straight forward |
| 21:32.20 | hads | Cool. |
| 21:32.29 | [andromeda] | I just want to pass variables into my AGI script, this is so frustrating.... |
| 21:32.45 | dlynes_laptop | hads, i don't have a problem with digium...just a problem with their hardware quality |
| 21:33.03 | hads | I know, we had this discussion already ;) |
| 21:33.04 | heffernan | dlynes_laptop: how is the price on sangoma cards compared to digium? |
| 21:33.07 | dlynes_laptop | hads, if their hardware quality was as good as sangoma's or even almost as good, i'd probably recommend them more often |
| 21:33.30 | dlynes_laptop | heffernan, retail pricing is pretty much the same |
| 21:33.37 | carl0s- | hi365: well. There's the Voiceblue 2N lite thingy.. whatever it's called. But.. there's a company called Portech in taiwan who have sort of 'spammed' a few of the voip forums offering their kit, but not many people who said they've used it. I checked out the user manual of the MT-370 and it looks perfect. You wouldn't beleive the price. Anyway there are one or two (I've counted two so far) Aussies who've used it, and had good results. $170 US + $40 |
| 21:33.37 | heffernan | k. |
| 21:34.07 | dlynes_laptop | heffernan, sangoma's driver install needs some user friendliness injected into it |
| 21:34.08 | heffernan | carl0s-: URL? |
| 21:34.11 | hads | The main problem in my country is that Digium cards have a telepermit and Sangoma don't. So I can't legally recommend Sangoma to be connected to the phone network :( |
| 21:34.16 | dlynes_laptop | heffernan, but the quality of the hardware is much better |
| 21:34.37 | dlynes_laptop | hads, oz? |
| 21:34.37 | *** part/#asterisk firejon (n=jellis@206-169-49-105.static.twtelecom.net) |
| 21:34.44 | hads | NZ |
| 21:34.47 | dlynes_laptop | ah |
| 21:34.48 | carl0s- | heffernan: they advertise them on eBay (ebay id portech_888), but the site is http://www.portech.com.tw . I'm very excited about this thing :D It's the MT-370 I've ordered. |
| 21:35.18 | hads | NZ's permitting is annoying. I don't have 20K to go for the permits myself. |
| 21:35.36 | dlynes_laptop | heh |
| 21:35.37 | dlynes_laptop | cute |
| 21:35.40 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 21:35.40 | dlynes_laptop | Within African communities, the term Sangoma is used to describe a man or woman skilled as diviner, spiritual healer, high priest, medicine man or woman and counsellor. Skilled at accessing and interacting with non-material dimensions, Sangomas can tap into hidden information, which can enhance the diagnosis of any condition be it health, legal or societal. |
| 21:36.23 | hi365 | carl0s- where did u find the sip-gsm gateway? |
| 21:36.33 | carl0s- | heffernan: sorry, it's the MV-370, not MT. |
| 21:37.03 | carl0s- | hi365: www.portech.com.tw . MV-370. the lady there.. saleschick Rebecca sells them on eBay as eBay ID portech_888 |
| 21:37.48 | *** part/#asterisk jm|home (n=jamiem@dsl-217-155-242-137.zen.co.uk) |
| 21:37.55 | dlynes_laptop | heh...8 means you'll get lots of money |
| 21:38.45 | carl0s- | hi365: http://cgi.ebay.co.uk/VoIP-GSM-Gateway_W0QQitemZ330008098301QQihZ014QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 21:39.00 | [andromeda] | Has anyone used AGI scripts? |
| 21:39.17 | carl0s- | hi365: the one I bought was on "Buy It Now" for $170 though. Mine was priced in Taiwanese Dollars ($5,500) |
| 21:39.47 | hi365 | wonder if its any good |
| 21:39.48 | Mercestes | Eh, why not? It's all made in Taiwan anyways. |
| 21:39.55 | hi365 | the price sounds prety decent |
| 21:40.08 | dlynes_laptop | hads, try http://www.sangoma.com/main/partners/resellers/au; maybe if you use the email address there, they might pay the $20K to get it certified in NZ |
| 21:40.22 | *** join/#asterisk tdonahue-laptop (n=tdonahue@www.vonworldwide.com) |
| 21:40.27 | carl0s- | hi365: pretty outrageously decent I'd say. I haven't priced the Voiceblue 2N lite or whatever it's called, but I suspect that'd be more like $700. |
| 21:41.03 | [andromeda] | can anyone even see what i'm typing? |
| 21:41.04 | carl0s- | plus the Voiceblue is dual-sim.. dual-channel. This is single-channel. |
| 21:41.08 | carl0s- | [andromeda]: yeh |
| 21:41.15 | sevard | [andromeda]: no |
| 21:41.31 | hads | dlynes_laptop: Yeah, I am setup to supply from them at the moment. They don't want to fork out either :/ |
| 21:41.47 | hads | Aparently it's quite an annoying process. |
| 21:42.01 | heffernan | how do i use a macro in a realtime extension? |
| 21:42.16 | Mercestes | realtime extension....eww. |
| 21:42.40 | heffernan | will 'Macro', 'internal,SIP/\${EXTEN},30,r' work for app and appdata? |
| 21:42.43 | heffernan | Mercestes: it's no good? |
| 21:42.44 | carl0s- | "ERROR: module Zaptel is in use by wctdm." (unloading Zaptel) that's what I got when rebooting the dead Asterisk box just then. |
| 21:42.47 | Mercestes | Realtime extension uses MySQL as an overglorified text editor. |
| 21:43.06 | hads | carl0s-: When it was going down? |
| 21:43.17 | carl0s- | hads: yes. I only just got round to rebooting it. |
| 21:43.26 | CoffeeIV | anyone here have spandsp working with asterisk 1.2.9.1 ? |
| 21:43.40 | hi365 | carl0s-: ok but htats x4, no? |
| 21:43.51 | hads | That just means that it tried to remove the zaptel module before the wctdm module. |
| 21:43.56 | carl0s- | hi365: what's x4? |
| 21:44.00 | heffernan | CoffeeIV: i have. |
| 21:44.03 | hi365 | times for chanels |
| 21:44.08 | hi365 | for=four |
| 21:44.18 | Mercestes | heffernan: Scrap the RTA extensions and just use the flatfile....it's nicer. |
| 21:44.29 | Mercestes | Of course I'm getting pretty anti-RTA at this point.... |
| 21:44.52 | Mercestes | sip.conf is tollerable.....anything beyond that is the devil. |
| 21:44.53 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 21:44.53 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:44.59 | carl0s- | hi365: yep but what is? The Voiceblue 2N lite is 2-channels. So it'll take two SIM cards and make two simultaneous inbound/outbound GSM calls, over different networks if you like. The Portech MV-370 is only single channel. But that's enough for me. |
| 21:45.03 | CoffeeIV | heffernan: any tricks ? which version of spandsp did you use -- the latest 0.0.3, or 0.0.2 ? |
| 21:45.21 | heffernan | Mercestes: that's a poor solution if you want to enable customers customizing their own dialplan. |
| 21:45.24 | heffernan | CoffeeIV: 0.0.2. |
| 21:45.28 | heffernan | CoffeeIV: worked perfectly. |
| 21:45.44 | carl0s- | hads: it's back up now. zap show channels works, and lists ztdummy and '4'. Is it normal for my only FXO module to be in the farthest socket on the TDM400 board? |
| 21:45.44 | Mercestes | heffernan: Why would you want customres customizing their own dialplan? |
| 21:46.00 | sevard | Mercestes: so you can break your box. |
| 21:46.11 | Mercestes | sevard: Word! |
| 21:46.19 | carl0s- | hi365: when my gsm<->sip gateway comes, I'll do a mini-review with some pictures and opinions and see if I can post it somewhere of use. |
| 21:46.39 | hads | carl0s-: Yeah, they normally come like that. I think it's a left over from an old revision where some of the FXO modules had problems in slot 1. |
| 21:46.40 | hi365 | k. where should we look? |
| 21:46.42 | Mercestes | Or I guess loop phones back on themselves more effieciently than ever..or better yet..extensions.conf back in on itself...>.> |
| 21:46.57 | hi365 | maybe voip-info.org? |
| 21:47.04 | CoffeeIV | heffernan: can you easily check to see if it was 0.0.2pre26, or a different one ? what OS are you using ? I am getting an asterisk crash when it hits RxFax() |
| 21:47.12 | carl0s- | hi365: yeh. that's a wiki site isn't it? I should be able to post there I think. |
| 21:47.25 | Mercestes | heffernan: Use RTA in sip.conf and then setup a php script to allow them to do basic features such as call forwarding and stuff. |
| 21:47.43 | carl0s- | hads: great, thanks. I'm going to call /dev/gf0 from my 7960 through the Asterisk box via the TDM400P and see what happens now. |
| 21:48.07 | Mercestes | heffernan: Set up a table that links a sip presence to an "alias" and allow them to foward that sip presence to a number so the can *basically* do some basic config options like, foward to a cell # or go straight to voicemail, etc. |
| 21:48.37 | carl0s- | ooh. i've got the asterisk ringer. now the BT ring tone.. |
| 21:48.39 | Mercestes | heffernan: And it doesn't require RTA extensions...just a call to PHP_AGI and php_agi outputs the dialplan you require. |
| 21:49.25 | carl0s- | woo. This TDM400P sounds perfect. |
| 21:49.30 | carl0s- | absolutely spot-on. |
| 21:49.39 | Mercestes | heffernen: For me and you, exten => Dial(sip/myphone,30) is ok....for end users, you should be looking at something more like, "Checkbox: Ring phone? (Y/N)" |
| 21:49.40 | hads | nice one carl0s- |
| 21:50.10 | carl0s- | I wonder if I get her to hang up on me, if it'll all go wrong again. |
| 21:51.12 | hads | carl0s-: Dial in with your mobile and hangup before you answer. See what happens. |
| 21:51.34 | Mercestes | NO! Don't! It'll set fire. |
| 21:51.44 | hads | heh |
| 21:51.53 | carl0s- | :D i'm trying to sound interested in her chit-chat here.. |
| 21:52.00 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:54.13 | carl0s- | that seemed fine. I got her to hang up on me and the phone responded properly. |
| 21:54.59 | hads | Still test it with your mobile as it could still be holding the line open on voicemail etc. |
| 21:56.08 | carl0s- | that didn't. I just called the land line from my mobile, it rang but the CLI went "Disconected from the Asterisk server". So Asterisk has died when I called in. This is the problem I had with the X100P. I'm sure my zaptel is screwed. I'm using a stupid 'up and running in 1hr" asterisk distribution and I suspect the newest update to it is bad. |
| 21:56.20 | *** part/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 21:56.36 | hi365 | it has bugs |
| 21:56.42 | hads | aaahhh. You're using one of those. |
| 21:56.57 | carl0s- | hads: sssh. ;) |
| 21:57.07 | hads | Try using $FAVOURITE_DISTRO plus Asterisk from sauce. |
| 21:57.11 | hi365 | i wasted over 20hr because of it |
| 21:57.29 | carl0s- | hi365: because of a particular bug, or what? |
| 21:57.50 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 21:58.42 | hi365 | yup. the kerel something spinlock.h |
| 21:59.06 | carl0s- | hi365: oh, that one's well known. You just change the rw_lock line. That prevents you compiling zaptel at all. |
| 21:59.31 | carl0s- | hi365: must've been frustrating for you though. |
| 21:59.54 | hi365 | i wish i can say "been"! at the moment its "IS"! |
| 22:00.42 | hads | ~centosbug |
| 22:00.43 | jbot | from memory, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 22:00.43 | Dovid | is there any way to make ge the images of the files that are on a polycom 601 or do i have to get it from polycom edit the files and then have the phone grab it ? |
| 22:01.05 | hi365 | but <burnproof> seems to be doing a superb job! |
| 22:01.58 | *** join/#asterisk AgiNamu (n=AgiNamu@207.155.175.7) |
| 22:02.41 | hads | dlynes_laptop: I just dropped Sangoma an email direct as I was dealing with their distributor before. I'll see what happens. |
| 22:03.42 | AgiNamu | Anyone know of any issues with SendDTMF on ZAP channels? Playtones works fine, but SendDTMF does not (nothing heard). |
| 22:04.57 | SplasPood | is it possible to get asterisk to always send my SIP phones the ANI info, even if the privacy "bit" is set? |
| 22:05.33 | [TK]D-Fender | Dovid : the latter |
| 22:06.48 | AgiNamu | Splas, I think you can do this via CallingPres command |
| 22:07.02 | AgiNamu | or SetCallerPres |
| 22:07.14 | AgiNamu | of course, Asterisk has to be receiving the ANI of course |
| 22:07.16 | *** join/#asterisk Trazz (n=traderz@65.114.86.29) |
| 22:07.18 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 22:07.32 | SplasPood | AgiNamu: I think that sets what's being presented when you make an outbound call, not how asterisk deals with it when sending an inbound call to a SIP phone |
| 22:07.34 | SplasPood | hrm |
| 22:07.35 | *** part/#asterisk Trazz (n=traderz@65.114.86.29) |
| 22:07.39 | SplasPood | i guess thats the same difference.. |
| 22:07.43 | AgiNamu | yea :) |
| 22:07.44 | [TK]D-Fender | hads : Whats your issue? |
| 22:08.08 | AgiNamu | But of course, your provider has to be sending you the call with presentation prohibited for Asterisk to take it out |
| 22:08.30 | hads | [TK]D-Fender: No issue, I just have a hard time selling Sangoma gear here (.nz) because it doesn't have a telepermit. |
| 22:08.40 | SplasPood | Nah see, when I *67 an inbound call, via my cell phone, the CDR gets the number and a name of 'anonymous' but the phones still see Unknown |
| 22:08.43 | sevard | bawhaha my boss ripped a big one |
| 22:08.46 | sevard | old dude |
| 22:09.05 | AgiNamu | oh ok cool |
| 22:09.05 | [TK]D-Fender | hads : Goota love the cert process.... slows NZ/AU to hell and a handbasket |
| 22:09.20 | hads | Doesn't it just. |
| 22:09.58 | hads | That plus if it's someone other thatn the manufacturer that "owns" the cert they can hold us all to ransom. |
| 22:10.23 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 22:12.35 | MikeJ[Laptop] | <PROTECTED> |
| 22:14.41 | AgiNamu | Does anyone know what modules have to be loaded for SendDTMF to work? |
| 22:14.51 | sevard | i disregard these jives about our equiptment, 007. |
| 22:15.05 | AgiNamu | I have res_features, res_indications, app_senddtmf |
| 22:15.58 | heffernan | what does WARNING[25019]: app_macro.c:149 macro_exec: No such context 'macro-internal,SIP/67417053' for macro 'internal,SIP/67417053' mean? |
| 22:17.17 | *** join/#asterisk System010 (n=jgargano@hide247.cybergnostic.com) |
| 22:17.59 | System010 | hi, does anyone have experience with connecting an Inter-Tel Axxess phone system to an asterisk server |
| 22:21.07 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 22:22.08 | mtgh | system010: Yes |
| 22:23.26 | System010 | mtgh, what is the best method to create a trunk between the two? |
| 22:24.20 | mtgh | system010: What are you trying to do, what hardware do you have for both systems? How many channels do you need? |
| 22:25.33 | System010 | I've written an app with agi, and I have assigned it to an extension. What I want is for customers to dial into our company phone system, an inter-tel axxess, and be able to reach the extensions on the asterisk server |
| 22:25.33 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 22:25.48 | System010 | we also want this to handle a high volume of calls |
| 22:25.49 | mtgh | How many at the same time? |
| 22:25.55 | System010 | upwards of 100 |
| 22:26.07 | mtgh | Then you will need a quad t1 card for asterisk |
| 22:26.21 | heffernan | how do i manuall set the context in a macro? |
| 22:26.39 | System010 | I can't do this through IP? |
| 22:26.52 | mtgh | Do you have the axxess ip card? |
| 22:26.56 | System010 | yes |
| 22:27.35 | mtgh | Ah then you can. Have intertel system register with asterisk |
| 22:27.50 | System010 | what protocol? |
| 22:28.54 | heffernan | okay, Set(CONTEXT()=foo) did not work. |
| 22:29.00 | mtgh | SIP |
| 22:29.05 | System010 | ok, thought so |
| 22:29.12 | *** join/#asterisk andrejkw (n=andrejkw@c-71-57-143-216.hsd1.fl.comcast.net) |
| 22:29.15 | andrejkw | Hey guys. |
| 22:29.22 | System010 | so i just configure asterisk as a sip server. |
| 22:29.33 | System010 | I only need to create one entry in the sip.conf, right? |
| 22:29.38 | mtgh | Yep. Then you pass the calls over to it. Yes only one |
| 22:29.41 | System010 | and then create a sip trunk |
| 22:29.47 | System010 | sweet |
| 22:29.51 | andrejkw | I can't get any phones plugged into my PAP2 to ring, I can use them to call out without a problem, but they simply won't ring. I have no clue what I am doing wrong. |
| 22:30.01 | mtgh | System010: where are you? (state) |
| 22:30.07 | System010 | ct |
| 22:30.18 | *** join/#asterisk pdtmobile (n=ptinsley@adsl-070-148-122-203.sip.bna.bellsouth.net) |
| 22:30.41 | andrejkw | can someone help, please? |
| 22:30.51 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 22:31.01 | rene- | hello |
| 22:31.15 | heffernan | anybody knows what WARNING[25019]: app_macro.c:149 macro_exec: No such context 'macro-internal,SIP/67417053' for macro 'internal,SIP/67417053' means? |
| 22:31.26 | rob0 | ~wikis |
| 22:31.28 | jbot | wikis is probably http://www.voip-info.org |
| 22:31.32 | rene- | what is the difference between the channel states ring and ringing ? |
| 22:31.36 | rob0 | andrejkw: ^^ |
| 22:31.42 | heffernan | rob0: nothing on that theree, unfortunately. |
| 22:32.02 | andrejkw | <rob0>: ? |
| 22:32.05 | pdtmobile | anybody had problems with operator out working in voicemail |
| 22:32.08 | rene- | and what is the difference between new state up and new channel? |
| 22:32.21 | *** join/#asterisk Trazz (n=traderz@65.114.86.29) |
| 22:32.23 | Hmmhesays | shut up sevard |
| 22:32.42 | rene- | and third: why does new channel gets reported twice always |
| 22:32.44 | *** part/#asterisk Trazz (n=traderz@65.114.86.29) |
| 22:32.48 | rob0 | andrejkw: Your PAP2 is a SIP peer. |
| 22:33.13 | andrejkw | Yes |
| 22:33.13 | rob0 | I'm sure there is specific information on it at the wiki. |
| 22:33.17 | *** join/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com) |
| 22:33.20 | andrejkw | I have it all setup. |
| 22:33.31 | andrejkw | It is regged with the Asterisk server and everything. |
| 22:33.34 | rob0 | but perhaps not correctly :) |
| 22:33.36 | andrejkw | I just can't makt it ring. |
| 22:33.42 | benjamin7062 | If I want ONE outbound pots line... what card should I buy? |
| 22:33.48 | pdtmobile | when I hit 0 to operator out it deletes the voicemail and hangs up. I would at least expect it to bitch about not finding the operator extension |
| 22:33.50 | rob0 | dialplan issues I guess |
| 22:34.04 | andrejkw | rob0: http://pastebin.ca/90676 |
| 22:34.16 | pdtmobile | I have all my voicemail boxes in the default context with exten => o,1,doostuff in the default context (the only thing) |
| 22:34.44 | heffernan | can anyone provide example of realtime extension that is using a macro? |
| 22:36.33 | mtgh | benjamin7062: check the wiki or digium |
| 22:36.53 | andrejkw | Can someone help? I can't get any phone on the PAP2 to ring. I can call out, but they won't ring. Here are my extensions.conf and sip.conf files: http://pastebin.ca/90676 |
| 22:37.07 | benjamin7062 | mtgh, Yeah, just seeing if people have suggestions... I 'thought' someone mentioned $20.00 in cost for a PRI card. TDM400 was a few hundred. |
| 22:37.17 | benjamin7062 | mtgh, thought perhaps someone might suggest a cheaper alternative. |
| 22:37.40 | benjamin7062 | s/PRI card/analog card/ |
| 22:37.54 | mtgh | 20 bucks is a xp100 clone. They are crap. You get what you pay for |
| 22:38.22 | rene- | analog card is 20~ish the quality is low, it is good for testing and if you are lucky you can use it for your home pbx project |
| 22:38.31 | benjamin7062 | I simply need an analog card and a single pots line for 911 calls. |
| 22:38.37 | rene- | froogle and ebay are your friends |
| 22:38.41 | benjamin7062 | as an alternative. |
| 22:38.51 | andrejkw | anyone, please? :'( I am going nuts. |
| 22:39.06 | docelmo | kewl can I watch you throw a fit? |
| 22:39.31 | rene- | andrejkw |
| 22:39.39 | rene- | pap2-line2@pap2-line2 wont work |
| 22:39.46 | rene- | you meant pap2-line2&pap2-line2 |
| 22:39.57 | andrejkw | oh, so replace @ with &? |
| 22:39.59 | rene- | or more likely |
| 22:40.04 | rene- | pap2-line1&pap2-line2 |
| 22:40.10 | benjamin7062 | I'll keep reading. Just looking for a $30.00 or $40.00 analog card with a single port that can be used for outbound pots. Didn't know if anyone had used one. I'll keep reading, thx |
| 22:40.20 | *** join/#asterisk Druken (i=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 22:40.38 | Druken | afternoon everyone... anyone have experince with the WIP300 ? |
| 22:40.48 | andrejkw | nope, still doesn't work :( |
| 22:40.50 | carl0s- | I think it would be nice if the Asterisk CLI allowed Cisco-style shortening of keywords. i.e. sh za st |
| 22:40.51 | Hmmhesays | somewhat |
| 22:40.54 | rene- | andrejkw: iff what you want is to ring both fxs interfaces of the sipura |
| 22:41.01 | andrejkw | no |
| 22:41.10 | andrejkw | I only want to ring pap-line2 |
| 22:41.20 | andrejkw | if i dial anything out of pap2-line1 |
| 22:41.26 | benjamin7062 | I wish they'd make the CLI standardize the 'show' commands... ie, show sip peers instead of sip show peers, etc |
| 22:41.42 | rene- | then |
| 22:42.24 | rene- | Dial(SIP/pap2-line2,10,tr) |
| 22:42.50 | E-bola | Hmm have anybody seen an integration of asterix with microsoft crm? |
| 22:43.04 | andrejkw | I get the busy tone |
| 22:43.12 | andrejkw | :( |
| 22:43.16 | benjamin7062 | E-bola, I think I saw a bounty of some sort for it; might be mistaken |
| 22:43.17 | mtgh | e-bola: we have done it |
| 22:43.44 | benjamin7062 | E-bola, it'd probably be trivial to write your own using the manager interface |
| 22:44.01 | rene- | are they both registered |
| 22:44.10 | E-bola | mtgh: i guess its quite straight forward with the asterix .net project? |
| 22:44.25 | mtgh | No, we did it with direct sql queries to the ms back end |
| 22:44.26 | E-bola | I just dont see a point re-inventing the wheel if somebody else already did it |
| 22:44.31 | mtgh | its not nice, but it works |
| 22:44.33 | E-bola | have any1 seen a public availeble code? |
| 22:44.50 | andrejkw | yes, rene |
| 22:46.13 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 22:46.13 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:48.12 | andrejkw | rene: oops, my bad, I am so sorry, the other line wasn't registered |
| 22:48.59 | heffernan | has anybody been running extensions that calls macros in realtime? |
| 22:49.05 | rob0 | FWIW, it would be nice if * supported the make O=/output/directory like the 2.6.x Linux kernels. I need to compile for 2 different $ARCH on the same machine. |
| 22:49.21 | E-bola | heffernan: i use a macro for all outgoing calls |
| 22:49.23 | E-bola | if thats what u mean |
| 22:49.32 | heffernan | E-bola: yeah, but an extension stored in realtime. |
| 22:49.57 | E-bola | im not familiar with realtime |
| 22:50.03 | heffernan | i have 'Macro', 'internal,SIP/${EXTEN}' for app and appdata. |
| 22:50.11 | heffernan | but seems like something is wrong. |
| 22:50.29 | heffernan | i get WARNING[25019]: app_macro.c:149 macro_exec: No such context 'macro-internal,SIP/67417053' for macro 'internal,SIP/67417053' |
| 22:50.37 | heffernan | E-bola: oh, okay. |
| 22:53.34 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 22:54.34 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 22:55.33 | rene- | andrejkw: dont worry shit happens |
| 22:55.37 | andrejkw | hmm |
| 22:55.42 | *** join/#asterisk froguz (n=xxxxx@pc-95-155-104-200.cm.vtr.net) |
| 22:55.43 | rene- | did it work? |
| 22:55.47 | andrejkw | I can't get incoming working with voicestick. |
| 22:55.59 | rene- | SIP? behind NAT? |
| 22:56.00 | andrejkw | Yes, I can have one line call the other line. |
| 22:56.07 | andrejkw | Yes, SIP behind NAT. |
| 22:56.24 | rene- | ew |
| 22:56.39 | andrejkw | I have an account with VoiceStick and for some reason I can call out, but can't receive calls. |
| 22:56.44 | rene- | no ring? no audio? |
| 22:56.44 | andrejkw | The phone doesn't ring. |
| 22:56.47 | froguz | somebody knows a good document for telephony reference? |
| 22:56.49 | andrejkw | No ring. |
| 22:57.46 | rene- | if your sip trunk is giving you DID (ANI) you must have an extension that matches your DID in the incoming section of your sip trunk or you can use the 's' extension |
| 22:58.30 | andrejkw | hmm |
| 22:58.45 | andrejkw | h/o let me post what I have, because I am not sure |
| 22:58.59 | carl0s- | how do I make a test call to my extension via the CLI again? |
| 23:00.51 | andrejkw | http://pastebin.ca/90696 |
| 23:05.38 | *** join/#asterisk [hC] (n=hardcore@66.119.176.4) |
| 23:05.58 | [hC] | Strom_C: hey, you alive? |
| 23:06.56 | rob0 | Grrr, I installed asterisk-addons, but MOH is still starting mpg123. |
| 23:08.37 | [hC] | Anyone know why asterisk would decide on an fxo line not to pick up the line and place a call? I have a system that works in my office, but as soon as it goes to a client site, it wont place calls. Ive checked hte line with a voltmeter and an analog phone and everything looks fine... |
| 23:08.48 | *** join/#asterisk sugardave (n=not@cpe-66-68-164-115.austin.res.rr.com) |
| 23:09.50 | *** join/#asterisk terrapen (i=cjs@saba.island.nu) |
| 23:10.20 | rob0 | And I still get "Request to schedule in the past" :( |
| 23:10.25 | terrapen | hrmm...does anybody know what would cause music-on-hold to start and then immediately stop? I'm using a custom app. |
| 23:10.46 | terrapen | My custom app runs just fine from the commandline and ouputs 8bit 8khz sound |
| 23:12.12 | andrejkw | Hmm what does "inbound routing" mean? |
| 23:12.20 | rene- | inbound routing is what i told you |
| 23:12.22 | *** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca) |
| 23:12.25 | rene- | it routes inbound calls |
| 23:12.36 | andrejkw | http://voxilla.com/index.php?name=PNphpBB2&file=viewtopic&p=43525#43525 |
| 23:12.53 | andrejkw | Here he says, 2) Go to inbound routing and setup a destination for your phone number. |
| 23:13.06 | terrapen | <PROTECTED> |
| 23:13.07 | terrapen | <PROTECTED> |
| 23:13.10 | terrapen | bastard! |
| 23:13.17 | terrapen | no indication of why it stopped |
| 23:13.20 | *** join/#asterisk HolyGod (i=nobody@got.securebinary.com) |
| 23:13.30 | rob0 | At the wiki I found: "load => app_dial.so ; Dialing Application - Requires res_features.so, res_musiconhold.so" ... is it true that app_dial requires MOH? Why? |
| 23:13.37 | rene- | andrejkw: you add a _ before 13.... |
| 23:13.51 | rene- | _ is needed whenever you want to make a match |
| 23:14.00 | rene- | you could use an exten 's' and get away with it |
| 23:14.04 | rene- | the issue is |
| 23:14.14 | rene- | do you see the call going into your machine? |
| 23:14.22 | rene- | because that could be happening |
| 23:14.24 | rob0 | Matchmaker matchmaker make me a match! Find me a find, catch me a catch! |
| 23:15.01 | andrejkw | rene-: How do I check if the call is even coming in? |
| 23:15.02 | terrapen | i swear, I've been using * for three years and have never been this stumped |
| 23:15.03 | [hC] | hrmf... why wont asterisk pick this line up and call out.. |
| 23:15.09 | terrapen | i have to figure out how to debug MoH |
| 23:15.21 | [hC] | anyone know how asterisk tests an fxo line to see wether or not its ready to accept outgoing calls? |
| 23:15.39 | rene- | andrewjkw: connect to the cli |
| 23:15.40 | terrapen | hc, well, it knows what channels are in use |
| 23:15.41 | *** join/#asterisk anthm (n=anthm@000-437-111.area4.spcsdns.net) |
| 23:15.41 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:15.44 | rene- | set verbose 5 |
| 23:16.04 | [hC] | terrapen: lets say if none are in use |
| 23:16.17 | andrejkw | rene-: huh? o.O |
| 23:16.22 | terrapen | heh, no amount of verbosity seems to be able to debug dying MoH |
| 23:16.22 | andrejkw | rene-: you just lost me |
| 23:16.30 | terrapen | andrew, go read the asterisk book |
| 23:16.57 | andrejkw | don't have the book. |
| 23:17.08 | terrapen | its free |
| 23:17.09 | terrapen | a PDF |
| 23:17.13 | drray | too bad the book is not available online |
| 23:17.14 | terrapen | go download it and read it. |
| 23:17.44 | andrejkw | ok, but can you tell me how to get into the CLI first? :P |
| 23:18.00 | drray | asterisk by itself, then asterisk -r |
| 23:18.06 | rob0 | For papa, make him a scholar; for mama, make him rich as a king! |
| 23:18.20 | anthm | Meet the author of the book and listen to him talk at CluecCon http://www.cluecon.com =D |
| 23:18.28 | *** join/#asterisk morex (i=morex@host86-133-105-226.range86-133.btcentralplus.com) |
| 23:18.30 | drray | build a man a fire, warm for a night.. set a man on fire.. warm the rest of his life. |
| 23:18.36 | morex | How do you force an exit with core dump in C? |
| 23:18.52 | anthm | abort(); |
| 23:18.55 | morex | Thx |
| 23:19.23 | anthm | np |
| 23:19.35 | *** join/#asterisk Kryojenik (n=DOH@24-107-84-247.dhcp.stls.mo.charter.com) |
| 23:19.58 | andrejkw | Nope, nothing is coming in. |
| 23:20.25 | Kryojenik | Anyone know where I can get an Israeli DID? |
| 23:20.57 | morex | Anybody know how to do a producer/consumer buffered writer in C? |
| 23:20.57 | anthm | run it in the foreground kill all instances and run asterisk -c "for console" |
| 23:22.14 | anthm | like a multithreaded FIFO queue? |
| 23:22.24 | andrejkw | THis is weird. |
| 23:22.32 | morex | Anthm: Something like that |
| 23:22.47 | morex | One thread to write to a buffer, another to write from that to the network... |
| 23:22.50 | rob0 | Okay, apparently app_dial.so does not need res_musiconhold.so. |
| 23:22.51 | andrejkw | I can call the number, I hear the ringing, but nothing even comes near asterisk. |
| 23:23.18 | rene- | nothing? |
| 23:23.20 | terrapen | fuck, there *has* to be a way to debug music on hold. |
| 23:23.21 | rene- | well do this |
| 23:23.24 | rene- | sip debug |
| 23:23.27 | anthm | what are you buffering ? |
| 23:23.31 | rene- | within the cli |
| 23:23.31 | terrapen | evidently, * is launching the MoH application and then it dies |
| 23:23.34 | morex | Manager events :-) |
| 23:23.44 | rene- | if i aint coming then you are not properly register |
| 23:23.48 | rene- | ed to your itsp |
| 23:23.50 | *** part/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 23:24.02 | morex | Currently sent from within locked code all over the place. Asterisk then hangs on write() and poll() while they're sent to the network |
| 23:24.03 | andrejkw | Ok, I did sip debug. |
| 23:24.16 | morex | OK if you're not too busy, but when things get hectic you get deadlocks all over the place |
| 23:24.17 | andrejkw | Do I try and call again? |
| 23:24.31 | morex | I'd fix it myself only I don't 'do' C any more... |
| 23:24.43 | morex | Took our call centre down three times today :-( |
| 23:24.57 | andrejkw | Ok yes. |
| 23:25.01 | anthm | yes that is an accurate account. |
| 23:25.05 | andrejkw | When I dialed bunch of stuff showed up. |
| 23:25.32 | morex | I figured there are two ways to solve: |
| 23:25.38 | andrejkw | And then it said Destroying call '......@.....'' |
| 23:25.45 | morex | 1) Move all manager_event() calls outside locked code, or |
| 23:26.01 | morex | 2) Buffer manager_event() writes, and have a separate thread write to the network. |
| 23:26.20 | morex | At least that way the manager_event() calls would return immediately, so long as the buffer is not full. |
| 23:26.38 | morex | But I don't know how to do it in C... |
| 23:27.21 | andrejkw | rene-: :\ are you there? |
| 23:27.35 | anthm | that's how i do mine. when events are fired they are added to a threadsafe queue where a event thread consumes them and delivers them in a backend monitor thread to registered consumers, if a consumer does anything complicated, it clones the event and adds it to another queue for it's own private handler thread. |
| 23:27.56 | morex | Anthm: Where are yours? |
| 23:28.19 | anthm | http://www.freeswitch.org |
| 23:29.12 | morex | Would you mind if I adapted some of your code to fix Asterisk? We've had loads of people complain about this (see bug report 6626 for example) |
| 23:29.33 | anthm | you would need to use apr |
| 23:29.37 | anthm | it's not license compatible |
| 23:29.43 | anthm | they only like code they own |
| 23:29.47 | morex | What's APR? |
| 23:29.54 | anthm | Apache Portable Runtime |
| 23:30.15 | anthm | the abstraction library that makes apache so cool |
| 23:30.15 | morex | Oh OK |
| 23:30.26 | morex | Yeah I keep meaning to add it to my Tomcat installs |
| 23:31.20 | morex | Ah well, perhaps it might be instructive as an example none the less. It can't be that hard - it's just that i'm so used to Java now where this kind of thing is so easy! |
| 23:32.35 | russellb | manager events are queued on to a master list in the trunk code |
| 23:32.44 | russellb | for the 1.4 release ... |
| 23:32.48 | morex | But not 1.2? |
| 23:32.52 | russellb | not 1.2 |
| 23:32.59 | morex | Maybe a backport is in order... |
| 23:33.13 | russellb | the changes are pretty significant |
| 23:33.25 | morex | Or maybe I should consider upgrading to trunk now... |
| 23:35.12 | morex | Sure is a pain getting shouted at when it dealocks! |
| 23:35.21 | morex | deadlocks even |
| 23:35.27 | russellb | have you tried the manager proxy? |
| 23:35.41 | russellb | that has helped a lot of people on 1.2 (and earlier) |
| 23:35.47 | morex | I can't see how it would help - I only have one manager client |
| 23:35.50 | russellb | ah. |
| 23:36.06 | anthm | are you sure it's events ? |
| 23:36.09 | andrejkw | Does anyone have VoiceStick setup with Asterisk? |
| 23:36.27 | morex | manager.c line 1511 (manager_event): '&sessionlock' was locked here |
| 23:36.43 | morex | I only very rarely send commands |
| 23:37.29 | *** join/#asterisk knarfly (n=knarf@c-69-180-98-189.hsd1.fl.comcast.net) |
| 23:37.42 | anthm | I did submit a patch a while ago to do internal manager event handlers |
| 23:38.04 | morex | What do you mean? |
| 23:38.34 | morex | Oh, like handling manager events from inside Asterisk? |
| 23:38.43 | anthm | with that you can make it bind to events from within C so instead of using the socket connection the function from your module gets called when an event happens |
| 23:38.59 | morex | No good to us - we have to do it over the network |
| 23:39.05 | anthm | which you can then translate to jabber or something |
| 23:39.08 | morex | Nice idea though |
| 23:39.16 | morex | Oh I C. |
| 23:39.22 | anthm | the point was you can bypass the default one and make another one |
| 23:39.27 | morex | Yeah OK |
| 23:39.31 | anthm | that does the backend thread business |
| 23:39.49 | morex | I'd probably just patch ast_carefulwrite not to block. |
| 23:40.06 | anthm | not blocking is why it's called carefulwrite iirc lol |
| 23:40.21 | morex | Nah it blocks on write() and poll() |
| 23:40.50 | morex | Could be wrong of course |
| 23:41.03 | *** join/#asterisk MoutaPT (n=MoutaPT@a83-132-239-175.cpe.netcabo.pt) |
| 23:41.27 | morex | Well if it's already there in 1.4 I might give it a go tomorrow. |
| 23:41.34 | MoutaPT | hi all, any one experienced with a2biliing? |
| 23:41.41 | anthm | mixing sockets and polling with threads is a tricky recipe |
| 23:41.48 | morex | yup |
| 23:41.59 | *** join/#asterisk mattfletcher (n=matt@mattspc.matt-fletcher.co.uk) |
| 23:42.28 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 23:43.01 | mattfletcher | I am having problems setting up port forwarding to my *. My router will only allow me to set up 500 port forwards (e.g.: 10000 to 10499). Can I make things work like this? |
| 23:43.28 | MoutaPT | tzafrir: Are you experienced with A2billing? |
| 23:43.36 | Strom_C | mattfletcher: consider using IAX or restricting the UDP ports that SIP uses |
| 23:43.48 | mattfletcher | Failing that, could I set up my * as a DHCP & NAT server and simply forward all traffic from the router to the *. I believe this is called a DMZ? |
| 23:43.49 | Strom_C | mattfletcher: or put asterisk on a public IP address |
| 23:43.57 | andrejkw | Does anyone have VoiceStick setup with Asterisk? |
| 23:44.10 | mattfletcher | it must be SIP as I'm using SIPGate and that is all it supports |
| 23:44.36 | mattfletcher | if i restrict the ports (rtp.conf correct?) do i need to inform sipgate of this somehow? |
| 23:44.53 | Strom_C | mattfletcher: no...SDP will handle that |
| 23:45.06 | mattfletcher | what is sdp? |
| 23:45.42 | Strom_C | session description protocol |
| 23:46.00 | Strom_C | handles port selection and codec negotiation during the setup of a SIP call |
| 23:46.03 | mattfletcher | ah ok |
| 23:46.33 | *** join/#asterisk adker (n=adker@70-100-230-148.br1.glv.ny.frontiernet.net) |
| 23:46.42 | mattfletcher | right, i have set it to use 10000 to 10400 (i could do with the other 100 for ssh, web and vnc etc) is that enough ports? |
| 23:47.01 | mattfletcher | presumably, once i've done that i just restart * and it might work? |
| 23:47.38 | Strom_C | you only need like four max per call |
| 23:48.01 | Strom_C | and thats assuming you're doing rfc2833 dtmf |
| 23:48.17 | andrejkw | I have a problem. For some reason I can't get Asterisk to receive calls from my SIP Provider (VoiceStick). I can make calls, but not receive. |
| 23:49.12 | mattfletcher | well that doesn't seem to have fixed it |
| 23:49.46 | mattfletcher | i'm still getting sipgate's voicemail message when i call my sipgate number (01524 888922 (uk)) |
| 23:50.13 | Strom_C | mattfletcher: put your asterisk box on a public IP address |
| 23:50.54 | carl0s- | mattfletcher: lol. just ringing your number for a laugh ;D |
| 23:50.59 | mattfletcher | okay, thanks for your help strom_c |
| 23:51.02 | *** join/#asterisk saftsack (n=saftsack@p54A7F566.dip.t-dialin.net) |
| 23:51.10 | rob0 | mattfletcher: won't your router accept a port range? |
| 23:51.12 | mattfletcher | please do carlos, what do you get? |
| 23:51.43 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 23:51.53 | carl0s- | mattfletcher: their voicemail. a really slow computer saying "0..1..5..2..4.....8....8..." then really fast bloke saying "isunavailablepleaseleaveamessage" |
| 23:52.26 | carl0s- | My Asterisk box isn't on a public IP. |
| 23:52.26 | carl0s- | oh |
| 23:52.46 | carl0s- | I'm with Sipgate too. although I have just reformatted my Asterisk box due to Zap problems. |
| 23:52.46 | mattfletcher | rob0: it is a cheap conexant router. i had to flash the firmware to get it to accept ranges, and now it will do it, but all it does is automatically create loads of individual entries. The hardware limit must be 500. |
| 23:53.13 | carl0s- | mattfletcher: I didn't forward any ports on my router (win2k3 RRAS) and sipgate worked fine. I followed the example in their help page. |
| 23:53.15 | mattfletcher | have you got it to work with port forwarding? |
| 23:53.24 | rob0 | mattfletcher: time to assemble a junky old Linux box. :) |
| 23:53.26 | mattfletcher | for incoming it worked fine? |
| 23:53.29 | carl0s- | yep |
| 23:53.35 | andrejkw | I have a problem. For some reason I can't get Asterisk to receive calls from my SIP Provider (VoiceStick). I can make calls, but not receive. |
| 23:53.54 | mattfletcher | rob0, what would I do with a junky old linux box (apart from run asterisk of course!) |
| 23:53.56 | andrejkw | Please, someone help :'( |
| 23:54.21 | rob0 | mattfletcher: replace the router? I use Linux machines as routers. |
| 23:54.26 | mattfletcher | andrejkw: sounds like ur suffering the same symptoms as me, take a look at the advice i've been given already |
| 23:54.27 | carl0s- | mattfletcher: turn it into a better router :). I wouldn't bother though. Those Conexant boxes work ok. |
| 23:54.52 | mattfletcher | if i make my own router, what software am i looking at? |
| 23:54.53 | andrejkw | mattfletcher: Did you figure out how to fix it? |
| 23:55.07 | mattfletcher | not yet andrejkw |
| 23:55.41 | saftsack | mog, hi |
| 23:56.04 | carl0s- | mattfletcher: have you looked at https://secure.sipgate.co.uk/faq/index.php?aktion=anzeigen&type=devices&rubrik=715#num7 |
| 23:56.05 | rob0 | Well, I just make my own iptables rules. If you don't know enough to do that there are many ready-made iptables scripts and rulesets to choose from. |
| 23:56.09 | mattfletcher | carl0s: where is the example from sipgate you read? i seem to remember seeing a few different ones, maybe i picked the wrong one |
| 23:56.13 | mattfletcher | ah :) |
| 23:56.24 | carl0s- | mattfletcher: :) I went off the first one.. |
| 23:56.44 | mattfletcher | ah i picked the second, lemme try that |
| 23:57.06 | mattfletcher | btw carlos, are u in the uk? just to check we are using a similar setup... |
| 23:57.16 | carl0s- | mattfletcher: yep. I'm in Manchester. |
| 23:57.35 | mattfletcher | ah, a fellow northerner. at one time even a lancastrian like myself |
| 23:57.41 | carl0s- | :D |
| 23:57.43 | carl0s- | where're you? |
| 23:57.47 | mattfletcher | morecambe |
| 23:58.21 | carl0s- | (poor at geography). that sounds far away. I've heard of Morecambe bay. isn't that where Fawlty Towers was set? |
| 23:58.51 | mattfletcher | that 1st config - does it do incoming calls as well as outgoing? |
| 23:58.56 | andrejkw | This is crazy, everything I tried and the phone still won't ring. |
| 23:58.58 | rob0 | lol @ Fawlty |
| 23:59.19 | carl0s- | mattfletcher: I tested it for incoming. Didn't test outgoing as I have no credit/money on file with them yet. |
| 23:59.40 | mattfletcher | fraid not, fawlty is set down south somewhere i think. but yes, it is on morecambe bay |